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gstreamer/subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c

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/* GStreamer
* Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
* <2006> Lutz Mueller <lutz at topfrose dot de>
* <2015> Jan Schmidt <jan at centricular dot com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/*
* Unless otherwise indicated, Source Code is licensed under MIT license.
* See further explanation attached in License Statement (distributed in the file
* LICENSE).
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
* of the Software, and to permit persons to whom the Software is furnished to do
* so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
/**
* SECTION:element-rtspclientsink
*
* Makes a connection to an RTSP server and send data via RTSP RECORD.
* rtspclientsink strictly follows RFC 2326
*
* RTSP supports transport over TCP or UDP in unicast or multicast mode. By
* default rtspclientsink will negotiate a connection in the following order:
* UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
* protocols can be controlled with the #GstRTSPClientSink:protocols property.
*
* rtspclientsink will internally instantiate an RTP session manager element
* that will handle the RTCP messages to and from the server, jitter removal,
* and packet reordering.
* This feature is implemented using the gstrtpbin element.
*
* rtspclientsink accepts any stream for which there is an installed payloader,
* creates the payloader and manages payload-types, as well as RTX setup.
* The new-payloader signal is fired when a payloader is created, in case
* an app wants to do custom configuration (such as for MTU).
*
* ## Example launch line
*
* |[
* gst-launch-1.0 videotestsrc ! jpegenc ! rtspclientsink location=rtsp://some.server/url
* ]| Establish a connection to an RTSP server and send JPEG encoded video packets
*/
/* FIXMEs
* - Handle EOS properly and shutdown. The problem with EOS is we don't know
* when the server has received all data, so we don't know when to do teardown.
* At the moment, we forward EOS to the app as soon as we stop sending. Is there
* a way to know from the receiver that it's got all data? Some session timeout?
* - Implement extension support for Real / WMS if they support RECORD?
* - Add support for network clock synchronised streaming?
* - Fix crypto key nego so SAVP/SAVPF profiles work.
* - Test (&fix?) HTTP tunnel support
* - Add an address pool object for GstRTSPStreams to use for multicast
* - Test multicast UDP transport
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#ifdef HAVE_UNISTD_H
#include <unistd.h>
#endif /* HAVE_UNISTD_H */
#include <stdlib.h>
#include <string.h>
#include <stdio.h>
#include <stdarg.h>
#include <gst/net/gstnet.h>
#include <gst/sdp/gstsdpmessage.h>
#include <gst/sdp/gstmikey.h>
#include <gst/rtp/rtp.h>
#include "gstrtspclientsink.h"
typedef struct _GstRtspClientSinkPad GstRtspClientSinkPad;
typedef GstGhostPadClass GstRtspClientSinkPadClass;
struct _GstRtspClientSinkPad
{
GstGhostPad parent;
GstElement *custom_payloader;
guint ulpfec_percentage;
};
enum
{
PROP_PAD_0,
PROP_PAD_PAYLOADER,
PROP_PAD_ULPFEC_PERCENTAGE
};
#define DEFAULT_PAD_ULPFEC_PERCENTAGE 0
static GType gst_rtsp_client_sink_pad_get_type (void);
G_DEFINE_TYPE (GstRtspClientSinkPad, gst_rtsp_client_sink_pad,
GST_TYPE_GHOST_PAD);
#define GST_TYPE_RTSP_CLIENT_SINK_PAD (gst_rtsp_client_sink_pad_get_type ())
#define GST_RTSP_CLIENT_SINK_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSP_CLIENT_SINK_PAD,GstRtspClientSinkPad))
static void
gst_rtsp_client_sink_pad_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtspClientSinkPad *pad;
pad = GST_RTSP_CLIENT_SINK_PAD (object);
switch (prop_id) {
case PROP_PAD_PAYLOADER:
GST_OBJECT_LOCK (pad);
if (pad->custom_payloader)
gst_object_unref (pad->custom_payloader);
pad->custom_payloader = g_value_get_object (value);
gst_object_ref_sink (pad->custom_payloader);
GST_OBJECT_UNLOCK (pad);
break;
case PROP_PAD_ULPFEC_PERCENTAGE:
GST_OBJECT_LOCK (pad);
pad->ulpfec_percentage = g_value_get_uint (value);
GST_OBJECT_UNLOCK (pad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtsp_client_sink_pad_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtspClientSinkPad *pad;
pad = GST_RTSP_CLIENT_SINK_PAD (object);
switch (prop_id) {
case PROP_PAD_PAYLOADER:
GST_OBJECT_LOCK (pad);
g_value_set_object (value, pad->custom_payloader);
GST_OBJECT_UNLOCK (pad);
break;
case PROP_PAD_ULPFEC_PERCENTAGE:
GST_OBJECT_LOCK (pad);
g_value_set_uint (value, pad->ulpfec_percentage);
GST_OBJECT_UNLOCK (pad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtsp_client_sink_pad_dispose (GObject * object)
{
GstRtspClientSinkPad *pad = GST_RTSP_CLIENT_SINK_PAD (object);
if (pad->custom_payloader)
gst_object_unref (pad->custom_payloader);
G_OBJECT_CLASS (gst_rtsp_client_sink_pad_parent_class)->dispose (object);
}
static void
gst_rtsp_client_sink_pad_class_init (GstRtspClientSinkPadClass * klass)
{
GObjectClass *gobject_klass;
gobject_klass = (GObjectClass *) klass;
gobject_klass->set_property = gst_rtsp_client_sink_pad_set_property;
gobject_klass->get_property = gst_rtsp_client_sink_pad_get_property;
gobject_klass->dispose = gst_rtsp_client_sink_pad_dispose;
g_object_class_install_property (gobject_klass, PROP_PAD_PAYLOADER,
g_param_spec_object ("payloader", "Payloader",
"The payloader element to use (NULL = default automatically selected)",
GST_TYPE_ELEMENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, PROP_PAD_ULPFEC_PERCENTAGE,
g_param_spec_uint ("ulpfec-percentage", "ULPFEC percentage",
"The percentage of ULP redundancy to apply", 0, 100,
DEFAULT_PAD_ULPFEC_PERCENTAGE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_rtsp_client_sink_pad_init (GstRtspClientSinkPad * pad)
{
}
static GstPad *
gst_rtsp_client_sink_pad_new (const GstPadTemplate * pad_tmpl,
const gchar * name)
{
GstRtspClientSinkPad *ret;
ret =
g_object_new (GST_TYPE_RTSP_CLIENT_SINK_PAD, "direction", GST_PAD_SINK,
"template", pad_tmpl, "name", name, NULL);
return GST_PAD (ret);
}
GST_DEBUG_CATEGORY_STATIC (rtsp_client_sink_debug);
#define GST_CAT_DEFAULT (rtsp_client_sink_debug)
static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS_ANY); /* Actual caps come from available set of payloaders */
enum
{
SIGNAL_HANDLE_REQUEST,
SIGNAL_NEW_MANAGER,
SIGNAL_NEW_PAYLOADER,
SIGNAL_REQUEST_RTCP_KEY,
SIGNAL_ACCEPT_CERTIFICATE,
SIGNAL_UPDATE_SDP,
LAST_SIGNAL
};
enum _GstRTSPClientSinkNtpTimeSource
{
NTP_TIME_SOURCE_NTP,
NTP_TIME_SOURCE_UNIX,
NTP_TIME_SOURCE_RUNNING_TIME,
NTP_TIME_SOURCE_CLOCK_TIME
};
#define GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE (gst_rtsp_client_sink_ntp_time_source_get_type())
static GType
gst_rtsp_client_sink_ntp_time_source_get_type (void)
{
static GType ntp_time_source_type = 0;
static const GEnumValue ntp_time_source_values[] = {
{NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"},
{NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"},
{NTP_TIME_SOURCE_RUNNING_TIME,
"Running time based on pipeline clock",
"running-time"},
{NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"},
{0, NULL, NULL},
};
if (!ntp_time_source_type) {
ntp_time_source_type =
g_enum_register_static ("GstRTSPClientSinkNtpTimeSource",
ntp_time_source_values);
}
return ntp_time_source_type;
}
#define DEFAULT_LOCATION NULL
#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
#define DEFAULT_DEBUG FALSE
#define DEFAULT_RETRY 20
#define DEFAULT_TIMEOUT 5000000
#define DEFAULT_UDP_BUFFER_SIZE 0x80000
#define DEFAULT_TCP_TIMEOUT 20000000
#define DEFAULT_LATENCY_MS 2000
#define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE
#define DEFAULT_PROXY NULL
#define DEFAULT_RTP_BLOCKSIZE 0
#define DEFAULT_USER_ID NULL
#define DEFAULT_USER_PW NULL
#define DEFAULT_PORT_RANGE NULL
#define DEFAULT_UDP_RECONNECT TRUE
#define DEFAULT_MULTICAST_IFACE NULL
#define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
#define DEFAULT_TLS_DATABASE NULL
#define DEFAULT_TLS_INTERACTION NULL
#define DEFAULT_NTP_TIME_SOURCE NTP_TIME_SOURCE_NTP
#define DEFAULT_USER_AGENT "GStreamer/" PACKAGE_VERSION
#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
#define DEFAULT_RTX_TIME_MS 500
#define DEFAULT_PUBLISH_CLOCK_MODE GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK
enum
{
PROP_0,
PROP_LOCATION,
PROP_PROTOCOLS,
PROP_DEBUG,
PROP_RETRY,
PROP_TIMEOUT,
PROP_TCP_TIMEOUT,
PROP_LATENCY,
PROP_RTX_TIME,
PROP_DO_RTSP_KEEP_ALIVE,
PROP_PROXY,
PROP_PROXY_ID,
PROP_PROXY_PW,
PROP_RTP_BLOCKSIZE,
PROP_USER_ID,
PROP_USER_PW,
PROP_PORT_RANGE,
PROP_UDP_BUFFER_SIZE,
PROP_UDP_RECONNECT,
PROP_MULTICAST_IFACE,
PROP_SDES,
PROP_TLS_VALIDATION_FLAGS,
PROP_TLS_DATABASE,
PROP_TLS_INTERACTION,
PROP_NTP_TIME_SOURCE,
PROP_USER_AGENT,
PROP_PROFILES,
PROP_PUBLISH_CLOCK_MODE,
};
static void gst_rtsp_client_sink_finalize (GObject * object);
static void gst_rtsp_client_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_client_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstClock *gst_rtsp_client_sink_provide_clock (GstElement * element);
static void gst_rtsp_client_sink_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static gboolean gst_rtsp_client_sink_set_proxy (GstRTSPClientSink * rtsp,
const gchar * proxy);
static void gst_rtsp_client_sink_set_tcp_timeout (GstRTSPClientSink *
rtsp_client_sink, guint64 timeout);
static GstStateChangeReturn gst_rtsp_client_sink_change_state (GstElement *
element, GstStateChange transition);
static void gst_rtsp_client_sink_handle_message (GstBin * bin,
GstMessage * message);
static gboolean gst_rtsp_client_sink_setup_auth (GstRTSPClientSink * sink,
GstRTSPMessage * response);
static gboolean gst_rtsp_client_sink_loop_send_cmd (GstRTSPClientSink * sink,
gint cmd, gint mask);
static GstRTSPResult gst_rtsp_client_sink_open (GstRTSPClientSink * sink,
gboolean async);
static GstRTSPResult gst_rtsp_client_sink_record (GstRTSPClientSink * sink,
gboolean async);
static GstRTSPResult gst_rtsp_client_sink_pause (GstRTSPClientSink * sink,
gboolean async);
static GstRTSPResult gst_rtsp_client_sink_close (GstRTSPClientSink * sink,
gboolean async, gboolean only_close);
static gboolean gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink);
static gboolean gst_rtsp_client_sink_uri_set_uri (GstURIHandler * handler,
const gchar * uri, GError ** error);
static gchar *gst_rtsp_client_sink_uri_get_uri (GstURIHandler * handler);
static gboolean gst_rtsp_client_sink_loop (GstRTSPClientSink * sink);
static void gst_rtsp_client_sink_connection_flush (GstRTSPClientSink * sink,
gboolean flush);
static GstPad *gst_rtsp_client_sink_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
static void gst_rtsp_client_sink_release_pad (GstElement * element,
GstPad * pad);
/* commands we send to out loop to notify it of events */
#define CMD_OPEN (1 << 0)
#define CMD_RECORD (1 << 1)
#define CMD_PAUSE (1 << 2)
#define CMD_CLOSE (1 << 3)
#define CMD_WAIT (1 << 4)
#define CMD_RECONNECT (1 << 5)
#define CMD_LOOP (1 << 6)
/* mask for all commands */
#define CMD_ALL ((CMD_LOOP << 1) - 1)
#define GST_ELEMENT_PROGRESS(el, type, code, text) \
G_STMT_START { \
gchar *__txt = _gst_element_error_printf text; \
gst_element_post_message (GST_ELEMENT_CAST (el), \
gst_message_new_progress (GST_OBJECT_CAST (el), \
GST_PROGRESS_TYPE_ ##type, code, __txt)); \
g_free (__txt); \
} G_STMT_END
static guint gst_rtsp_client_sink_signals[LAST_SIGNAL] = { 0 };
/*********************************
* GstChildProxy implementation *
*********************************/
static GObject *
gst_rtsp_client_sink_child_proxy_get_child_by_index (GstChildProxy *
child_proxy, guint index)
{
GObject *obj;
GstRTSPClientSink *cs = GST_RTSP_CLIENT_SINK (child_proxy);
GST_OBJECT_LOCK (cs);
if ((obj = g_list_nth_data (GST_ELEMENT (cs)->sinkpads, index)))
g_object_ref (obj);
GST_OBJECT_UNLOCK (cs);
return obj;
}
static guint
gst_rtsp_client_sink_child_proxy_get_children_count (GstChildProxy *
child_proxy)
{
guint count = 0;
GST_OBJECT_LOCK (child_proxy);
count = GST_ELEMENT (child_proxy)->numsinkpads;
GST_OBJECT_UNLOCK (child_proxy);
GST_INFO_OBJECT (child_proxy, "Children Count: %d", count);
return count;
}
static void
gst_rtsp_client_sink_child_proxy_init (gpointer g_iface, gpointer iface_data)
{
GstChildProxyInterface *iface = g_iface;
GST_INFO ("intializing child proxy interface");
iface->get_child_by_index =
gst_rtsp_client_sink_child_proxy_get_child_by_index;
iface->get_children_count =
gst_rtsp_client_sink_child_proxy_get_children_count;
}
#define gst_rtsp_client_sink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstRTSPClientSink, gst_rtsp_client_sink, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
gst_rtsp_client_sink_uri_handler_init);
G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY,
gst_rtsp_client_sink_child_proxy_init);
);
#ifndef GST_DISABLE_GST_DEBUG
static inline const gchar *
cmd_to_string (guint cmd)
{
switch (cmd) {
case CMD_OPEN:
return "OPEN";
case CMD_RECORD:
return "RECORD";
case CMD_PAUSE:
return "PAUSE";
case CMD_CLOSE:
return "CLOSE";
case CMD_WAIT:
return "WAIT";
case CMD_RECONNECT:
return "RECONNECT";
case CMD_LOOP:
return "LOOP";
}
return "unknown";
}
#endif
static void
gst_rtsp_client_sink_class_init (GstRTSPClientSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBinClass *gstbin_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbin_class = (GstBinClass *) klass;
GST_DEBUG_CATEGORY_INIT (rtsp_client_sink_debug, "rtspclientsink", 0,
"RTSP sink element");
gobject_class->set_property = gst_rtsp_client_sink_set_property;
gobject_class->get_property = gst_rtsp_client_sink_get_property;
gobject_class->finalize = gst_rtsp_client_sink_finalize;
g_object_class_install_property (gobject_class, PROP_LOCATION,
g_param_spec_string ("location", "RTSP Location",
"Location of the RTSP url to read",
DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
g_param_spec_flags ("protocols", "Protocols",
"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PROFILES,
g_param_spec_flags ("profiles", "Profiles",
"Allowed RTSP profiles", GST_TYPE_RTSP_PROFILE,
DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DEBUG,
g_param_spec_boolean ("debug", "Debug",
"Dump request and response messages to stdout",
DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RETRY,
g_param_spec_uint ("retry", "Retry",
"Max number of retries when allocating RTP ports.",
0, G_MAXUINT16, DEFAULT_RETRY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TIMEOUT,
g_param_spec_uint64 ("timeout", "Timeout",
"Retry TCP transport after UDP timeout microseconds (0 = disabled)",
0, G_MAXUINT64, DEFAULT_TIMEOUT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
"Fail after timeout microseconds on TCP connections (0 = disabled)",
0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LATENCY,
g_param_spec_uint ("latency", "Buffer latency in ms",
"Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTX_TIME,
g_param_spec_uint ("rtx-time", "Retransmission buffer in ms",
"Amount of ms to buffer for retransmission. 0 disables retransmission",
0, G_MAXUINT, DEFAULT_RTX_TIME_MS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:do-rtsp-keep-alive:
*
* Enable RTSP keep alive support. Some old server don't like RTSP
* keep alive and then this property needs to be set to FALSE.
*/
g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
"Send RTSP keep alive packets, disable for old incompatible server.",
DEFAULT_DO_RTSP_KEEP_ALIVE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:proxy:
*
* Set the proxy parameters. This has to be a string of the format
* [http://][user:passwd@]host[:port].
*/
g_object_class_install_property (gobject_class, PROP_PROXY,
g_param_spec_string ("proxy", "Proxy",
"Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:proxy-id:
*
* Sets the proxy URI user id for authentication. If the URI set via the
* "proxy" property contains a user-id already, that will take precedence.
*
*/
g_object_class_install_property (gobject_class, PROP_PROXY_ID,
g_param_spec_string ("proxy-id", "proxy-id",
"HTTP proxy URI user id for authentication", "",
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:proxy-pw:
*
* Sets the proxy URI password for authentication. If the URI set via the
* "proxy" property contains a password already, that will take precedence.
*
*/
g_object_class_install_property (gobject_class, PROP_PROXY_PW,
g_param_spec_string ("proxy-pw", "proxy-pw",
"HTTP proxy URI user password for authentication", "",
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:rtp-blocksize:
*
* RTP package size to suggest to server.
*/
g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
"RTP package size to suggest to server (0 = disabled)",
0, 65536, DEFAULT_RTP_BLOCKSIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_USER_ID,
g_param_spec_string ("user-id", "user-id",
"RTSP location URI user id for authentication", DEFAULT_USER_ID,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_USER_PW,
g_param_spec_string ("user-pw", "user-pw",
"RTSP location URI user password for authentication", DEFAULT_USER_PW,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:port-range:
*
* Configure the client port numbers that can be used to receive
* RTCP.
*/
g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
g_param_spec_string ("port-range", "Port range",
"Client port range that can be used to receive RTCP data, "
"eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:udp-buffer-size:
*
* Size of the kernel UDP receive buffer in bytes.
*/
g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
"Size of the kernel UDP receive buffer in bytes, 0=default",
0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
"Reconnect to the server if RTSP connection is closed when doing UDP",
DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
g_param_spec_string ("multicast-iface", "Multicast Interface",
"The network interface on which to join the multicast group",
DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES,
g_param_spec_boxed ("sdes", "SDES",
"The SDES items of this session",
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:tls-validation-flags:
*
* TLS certificate validation flags used to validate server
* certificate.
*
* GLib guarantees that if certificate verification fails, at least one
* error will be set, but it does not guarantee that all possible errors
* will be set. Accordingly, you may not safely decide to ignore any
* particular type of error.
*
* For example, it would be incorrect to mask %G_TLS_CERTIFICATE_EXPIRED if
* you want to allow expired certificates, because this could potentially be
* the only error flag set even if other problems exist with the
* certificate.
*
*/
g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
"TLS certificate validation flags used to validate the server certificate",
G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:tls-database:
*
* TLS database with anchor certificate authorities used to validate
* the server certificate.
*
*/
g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
g_param_spec_object ("tls-database", "TLS database",
"TLS database with anchor certificate authorities used to validate the server certificate",
G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:tls-interaction:
*
* A #GTlsInteraction object to be used when the connection or certificate
* database need to interact with the user. This will be used to prompt the
* user for passwords where necessary.
*
*/
g_object_class_install_property (gobject_class, PROP_TLS_INTERACTION,
g_param_spec_object ("tls-interaction", "TLS interaction",
"A GTlsInteraction object to prompt the user for password or certificate",
G_TYPE_TLS_INTERACTION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:ntp-time-source:
*
* allows to select the time source that should be used
* for the NTP time in outgoing packets
*
*/
g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
g_param_spec_enum ("ntp-time-source", "NTP Time Source",
"NTP time source for RTCP packets",
GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE, DEFAULT_NTP_TIME_SOURCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:user-agent:
*
* The string to set in the User-Agent header.
*
*/
g_object_class_install_property (gobject_class, PROP_USER_AGENT,
g_param_spec_string ("user-agent", "User Agent",
"The User-Agent string to send to the server",
DEFAULT_USER_AGENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:publish-clock-mode:
*
* Sets if and how the media clock should be published according to RFC7273.
*
* Since: 1.22
*
*/
g_object_class_install_property (gobject_class, PROP_PUBLISH_CLOCK_MODE,
g_param_spec_enum ("publish-clock-mode", "Publish Clock Mode",
"Clock publishing mode according to RFC7273",
GST_TYPE_RTSP_PUBLISH_CLOCK_MODE, DEFAULT_PUBLISH_CLOCK_MODE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink::handle-request:
* @rtsp_client_sink: a #GstRTSPClientSink
* @request: a #GstRTSPMessage
* @response: a #GstRTSPMessage
*
* Handle a server request in @request and prepare @response.
*
* This signal is called from the streaming thread, you should therefore not
* do any state changes on @rtsp_client_sink because this might deadlock. If you want
* to modify the state as a result of this signal, post a
* #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
* in some other way.
*
*/
gst_rtsp_client_sink_signals[SIGNAL_HANDLE_REQUEST] =
g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
0, NULL, NULL, NULL, G_TYPE_NONE, 2,
GST_TYPE_RTSP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE,
GST_TYPE_RTSP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
/**
* GstRTSPClientSink::new-manager:
* @rtsp_client_sink: a #GstRTSPClientSink
* @manager: a #GstElement
*
* Emitted after a new manager (like rtpbin) was created and the default
* properties were configured.
*
*/
gst_rtsp_client_sink_signals[SIGNAL_NEW_MANAGER] =
g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST, 0, NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
/**
* GstRTSPClientSink::new-payloader:
* @rtsp_client_sink: a #GstRTSPClientSink
* @payloader: a #GstElement
*
* Emitted after a new RTP payloader was created and the default
* properties were configured.
*
*/
gst_rtsp_client_sink_signals[SIGNAL_NEW_PAYLOADER] =
g_signal_new_class_handler ("new-payloader", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST, 0, NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
/**
* GstRTSPClientSink::request-rtcp-key:
* @rtsp_client_sink: a #GstRTSPClientSink
* @num: the stream number
*
* Signal emitted to get the crypto parameters relevant to the RTCP
* stream. User should provide the key and the RTCP encryption ciphers
* and authentication, and return them wrapped in a GstCaps.
*
*/
gst_rtsp_client_sink_signals[SIGNAL_REQUEST_RTCP_KEY] =
g_signal_new ("request-rtcp-key", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstRTSPClientSink::accept-certificate:
* @rtsp_client_sink: a #GstRTSPClientSink
* @peer_cert: the peer's #GTlsCertificate
* @errors: the problems with @peer_cert
* @user_data: user data set when the signal handler was connected.
*
* This will directly map to #GTlsConnection 's "accept-certificate"
* signal and be performed after the default checks of #GstRTSPConnection
* (checking against the #GTlsDatabase with the given #GTlsCertificateFlags)
* have failed. If no #GTlsDatabase is set on this connection, only this
* signal will be emitted.
*
* Since: 1.14
*/
gst_rtsp_client_sink_signals[SIGNAL_ACCEPT_CERTIFICATE] =
g_signal_new ("accept-certificate", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, g_signal_accumulator_true_handled, NULL, NULL,
G_TYPE_BOOLEAN, 3, G_TYPE_TLS_CONNECTION, G_TYPE_TLS_CERTIFICATE,
G_TYPE_TLS_CERTIFICATE_FLAGS);
/**
* GstRTSPClientSink::update-sdp:
* @rtsp_client_sink: a #GstRTSPClientSink
* @sdp: a #GstSDPMessage
*
* Emitted right before the ANNOUNCE request is sent to the server with the
* generated SDP. The SDP can be updated from signal handlers but the order
* and number of medias must not be changed.
*
* Since: 1.20
*/
gst_rtsp_client_sink_signals[SIGNAL_UPDATE_SDP] =
g_signal_new_class_handler ("update-sdp", G_TYPE_FROM_CLASS (klass),
0, 0, NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
gstelement_class->provide_clock = gst_rtsp_client_sink_provide_clock;
gstelement_class->change_state = gst_rtsp_client_sink_change_state;
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_request_new_pad);
gstelement_class->release_pad =
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_release_pad);
gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
&rtptemplate, GST_TYPE_RTSP_CLIENT_SINK_PAD);
gst_element_class_set_static_metadata (gstelement_class,
"RTSP RECORD client", "Sink/Network",
"Send data over the network via RTSP RECORD(RFC 2326)",
"Jan Schmidt <jan@centricular.com>");
gstbin_class->handle_message = gst_rtsp_client_sink_handle_message;
gst_type_mark_as_plugin_api (GST_TYPE_RTSP_CLIENT_SINK_PAD, 0);
gst_type_mark_as_plugin_api (GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE, 0);
}
static void
gst_rtsp_client_sink_init (GstRTSPClientSink * sink)
{
sink->conninfo.location = g_strdup (DEFAULT_LOCATION);
sink->protocols = DEFAULT_PROTOCOLS;
sink->debug = DEFAULT_DEBUG;
sink->retry = DEFAULT_RETRY;
sink->udp_timeout = DEFAULT_TIMEOUT;
gst_rtsp_client_sink_set_tcp_timeout (sink, DEFAULT_TCP_TIMEOUT);
sink->latency = DEFAULT_LATENCY_MS;
sink->rtx_time = DEFAULT_RTX_TIME_MS;
sink->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
gst_rtsp_client_sink_set_proxy (sink, DEFAULT_PROXY);
sink->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
sink->user_id = g_strdup (DEFAULT_USER_ID);
sink->user_pw = g_strdup (DEFAULT_USER_PW);
sink->client_port_range.min = 0;
sink->client_port_range.max = 0;
sink->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
sink->udp_reconnect = DEFAULT_UDP_RECONNECT;
sink->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
sink->sdes = NULL;
sink->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
sink->tls_database = DEFAULT_TLS_DATABASE;
sink->tls_interaction = DEFAULT_TLS_INTERACTION;
sink->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
sink->user_agent = g_strdup (DEFAULT_USER_AGENT);
sink->publish_clock_mode = DEFAULT_PUBLISH_CLOCK_MODE;
sink->profiles = DEFAULT_PROFILES;
/* protects the streaming thread in interleaved mode or the polling
* thread in UDP mode. */
g_rec_mutex_init (&sink->stream_rec_lock);
/* protects our state changes from multiple invocations */
g_rec_mutex_init (&sink->state_rec_lock);
g_mutex_init (&sink->send_lock);
g_mutex_init (&sink->preroll_lock);
g_cond_init (&sink->preroll_cond);
sink->state = GST_RTSP_STATE_INVALID;
g_mutex_init (&sink->conninfo.send_lock);
g_mutex_init (&sink->conninfo.recv_lock);
g_mutex_init (&sink->block_streams_lock);
g_cond_init (&sink->block_streams_cond);
g_mutex_init (&sink->open_conn_lock);
g_cond_init (&sink->open_conn_cond);
sink->internal_bin = (GstBin *) gst_bin_new ("rtspbin");
g_object_set (sink->internal_bin, "async-handling", TRUE, NULL);
gst_element_set_locked_state (GST_ELEMENT_CAST (sink->internal_bin), TRUE);
gst_bin_add (GST_BIN (sink), GST_ELEMENT_CAST (sink->internal_bin));
sink->next_dyn_pt = 96;
gst_sdp_message_init (&sink->cursdp);
GST_OBJECT_FLAG_SET (sink, GST_ELEMENT_FLAG_SINK);
}
static void
gst_rtsp_client_sink_finalize (GObject * object)
{
GstRTSPClientSink *rtsp_client_sink;
rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
gst_sdp_message_uninit (&rtsp_client_sink->cursdp);
g_free (rtsp_client_sink->conninfo.location);
gst_rtsp_url_free (rtsp_client_sink->conninfo.url);
g_free (rtsp_client_sink->conninfo.url_str);
g_free (rtsp_client_sink->user_id);
g_free (rtsp_client_sink->user_pw);
g_free (rtsp_client_sink->multi_iface);
g_free (rtsp_client_sink->user_agent);
if (rtsp_client_sink->uri_sdp) {
gst_sdp_message_free (rtsp_client_sink->uri_sdp);
rtsp_client_sink->uri_sdp = NULL;
}
if (rtsp_client_sink->provided_clock)
gst_object_unref (rtsp_client_sink->provided_clock);
if (rtsp_client_sink->sdes)
gst_structure_free (rtsp_client_sink->sdes);
if (rtsp_client_sink->tls_database)
g_object_unref (rtsp_client_sink->tls_database);
if (rtsp_client_sink->tls_interaction)
g_object_unref (rtsp_client_sink->tls_interaction);
/* free locks */
g_rec_mutex_clear (&rtsp_client_sink->stream_rec_lock);
g_rec_mutex_clear (&rtsp_client_sink->state_rec_lock);
g_mutex_clear (&rtsp_client_sink->conninfo.send_lock);
g_mutex_clear (&rtsp_client_sink->conninfo.recv_lock);
g_mutex_clear (&rtsp_client_sink->send_lock);
g_mutex_clear (&rtsp_client_sink->preroll_lock);
g_cond_clear (&rtsp_client_sink->preroll_cond);
g_mutex_clear (&rtsp_client_sink->block_streams_lock);
g_cond_clear (&rtsp_client_sink->block_streams_cond);
g_mutex_clear (&rtsp_client_sink->open_conn_lock);
g_cond_clear (&rtsp_client_sink->open_conn_cond);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_payloader_filter_func (GstPluginFeature * feature, gpointer user_data)
{
GstElementFactory *factory = NULL;
const gchar *klass;
if (!GST_IS_ELEMENT_FACTORY (feature))
return FALSE;
factory = GST_ELEMENT_FACTORY (feature);
if (gst_plugin_feature_get_rank (feature) == GST_RANK_NONE)
return FALSE;
if (!gst_element_factory_list_is_type (factory,
GST_ELEMENT_FACTORY_TYPE_PAYLOADER))
return FALSE;
klass =
gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_KLASS);
if (strstr (klass, "Codec") == NULL)
return FALSE;
if (strstr (klass, "RTP") == NULL)
return FALSE;
return TRUE;
}
static gint
compare_ranks (GstPluginFeature * f1, GstPluginFeature * f2)
{
gint diff;
const gchar *rname1, *rname2;
GstRank rank1, rank2;
rname1 = gst_plugin_feature_get_name (f1);
rname2 = gst_plugin_feature_get_name (f2);
rank1 = gst_plugin_feature_get_rank (f1);
rank2 = gst_plugin_feature_get_rank (f2);
/* HACK: Prefer rtpmp4apay over rtpmp4gpay */
if (g_str_equal (rname1, "rtpmp4apay"))
rank1 = GST_RANK_SECONDARY + 1;
if (g_str_equal (rname2, "rtpmp4apay"))
rank2 = GST_RANK_SECONDARY + 1;
diff = rank2 - rank1;
if (diff != 0)
return diff;
diff = strcmp (rname2, rname1);
return diff;
}
static GList *
gst_rtsp_client_sink_get_factories (void)
{
static GList *payloader_factories = NULL;
if (g_once_init_enter (&payloader_factories)) {
GList *all_factories;
all_factories =
gst_registry_feature_filter (gst_registry_get (),
gst_rtp_payloader_filter_func, FALSE, NULL);
all_factories = g_list_sort (all_factories, (GCompareFunc) compare_ranks);
g_once_init_leave (&payloader_factories, all_factories);
}
return payloader_factories;
}
static GstCaps *
gst_rtsp_client_sink_get_payloader_caps (GstElementFactory * factory)
{
const GList *tmp;
GstCaps *caps = gst_caps_new_empty ();
for (tmp = gst_element_factory_get_static_pad_templates (factory);
tmp; tmp = g_list_next (tmp)) {
GstStaticPadTemplate *template = tmp->data;
if (template->direction == GST_PAD_SINK) {
GstCaps *static_caps = gst_static_pad_template_get_caps (template);
GST_LOG ("Found pad template %s on factory %s",
template->name_template, gst_plugin_feature_get_name (factory));
if (static_caps)
caps = gst_caps_merge (caps, static_caps);
/* Early out, any is absorbing */
if (gst_caps_is_any (caps))
goto out;
}
}
out:
return caps;
}
static GstCaps *
gst_rtsp_client_sink_get_all_payloaders_caps (void)
{
/* Cached caps result */
static GstCaps *ret;
if (g_once_init_enter (&ret)) {
GList *factories, *cur;
GstCaps *caps = gst_caps_new_empty ();
factories = gst_rtsp_client_sink_get_factories ();
for (cur = factories; cur != NULL; cur = g_list_next (cur)) {
GstElementFactory *factory = GST_ELEMENT_FACTORY (cur->data);
GstCaps *payloader_caps =
gst_rtsp_client_sink_get_payloader_caps (factory);
caps = gst_caps_merge (caps, payloader_caps);
/* Early out, any is absorbing */
if (gst_caps_is_any (caps))
goto out;
}
GST_MINI_OBJECT_FLAG_SET (caps, GST_MINI_OBJECT_FLAG_MAY_BE_LEAKED);
out:
g_once_init_leave (&ret, caps);
}
/* Return cached result */
return gst_caps_ref (ret);
}
static GstElement *
gst_rtsp_client_sink_make_payloader (GstCaps * caps)
{
GList *factories, *cur;
factories = gst_rtsp_client_sink_get_factories ();
for (cur = factories; cur != NULL; cur = g_list_next (cur)) {
GstElementFactory *factory = GST_ELEMENT_FACTORY (cur->data);
const GList *tmp;
for (tmp = gst_element_factory_get_static_pad_templates (factory);
tmp; tmp = g_list_next (tmp)) {
GstStaticPadTemplate *template = tmp->data;
if (template->direction == GST_PAD_SINK) {
GstCaps *static_caps = gst_static_pad_template_get_caps (template);
GstElement *payloader = NULL;
if (gst_caps_can_intersect (static_caps, caps)) {
GST_DEBUG ("caps %" GST_PTR_FORMAT " intersects with template %"
GST_PTR_FORMAT " for payloader %s", caps, static_caps,
gst_plugin_feature_get_name (factory));
payloader = gst_element_factory_create (factory, NULL);
}
gst_caps_unref (static_caps);
if (payloader)
return payloader;
}
}
}
return NULL;
}
static GstRTSPStream *
gst_rtsp_client_sink_create_stream (GstRTSPClientSink * sink,
GstRTSPStreamContext * context, GstElement * payloader, GstPad * pad)
{
GstRTSPStream *stream = NULL;
guint pt, aux_pt, ulpfec_pt;
GST_OBJECT_LOCK (sink);
g_object_get (G_OBJECT (payloader), "pt", &pt, NULL);
if (pt >= 96 && pt <= sink->next_dyn_pt) {
/* Payloader has a dynamic PT, but one that's already used */
/* FIXME: Create a caps->ptmap instead? */
pt = sink->next_dyn_pt;
if (pt > 127)
goto no_free_pt;
GST_DEBUG_OBJECT (sink, "Assigning pt %u to stream %d", pt, context->index);
sink->next_dyn_pt++;
} else {
GST_DEBUG_OBJECT (sink, "Keeping existing pt %u for stream %d",
pt, context->index);
}
aux_pt = sink->next_dyn_pt;
if (aux_pt > 127)
goto no_free_pt;
sink->next_dyn_pt++;
ulpfec_pt = sink->next_dyn_pt;
if (ulpfec_pt > 127)
goto no_free_pt;
sink->next_dyn_pt++;
GST_OBJECT_UNLOCK (sink);
g_object_set (G_OBJECT (payloader), "pt", pt, NULL);
stream = gst_rtsp_stream_new (context->index, payloader, pad);
gst_rtsp_stream_set_client_side (stream, TRUE);
gst_rtsp_stream_set_retransmission_time (stream,
(GstClockTime) (sink->rtx_time) * GST_MSECOND);
gst_rtsp_stream_set_protocols (stream, sink->protocols);
gst_rtsp_stream_set_profiles (stream, sink->profiles);
gst_rtsp_stream_set_retransmission_pt (stream, aux_pt);
gst_rtsp_stream_set_buffer_size (stream, sink->udp_buffer_size);
if (sink->rtp_blocksize > 0)
gst_rtsp_stream_set_mtu (stream, sink->rtp_blocksize);
gst_rtsp_stream_set_multicast_iface (stream, sink->multi_iface);
gst_rtsp_stream_set_ulpfec_pt (stream, ulpfec_pt);
gst_rtsp_stream_set_ulpfec_percentage (stream, context->ulpfec_percentage);
gst_rtsp_stream_set_publish_clock_mode (stream, sink->publish_clock_mode);
#if 0
if (priv->pool)
gst_rtsp_stream_set_address_pool (stream, priv->pool);
#endif
return stream;
no_free_pt:
GST_OBJECT_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, NO_SPACE_LEFT, (NULL),
("Ran out of dynamic payload types."));
return NULL;
}
static GstPadProbeReturn
handle_payloader_block (GstPad * pad, GstPadProbeInfo * info,
GstRTSPStreamContext * context)
{
GstRTSPClientSink *sink = context->parent;
GST_INFO_OBJECT (sink, "Block on pad %" GST_PTR_FORMAT, pad);
g_mutex_lock (&sink->preroll_lock);
context->prerolled = TRUE;
g_cond_broadcast (&sink->preroll_cond);
g_mutex_unlock (&sink->preroll_lock);
GST_INFO_OBJECT (sink, "Announced preroll on pad %" GST_PTR_FORMAT, pad);
return GST_PAD_PROBE_OK;
}
static gboolean
gst_rtsp_client_sink_setup_payloader (GstRTSPClientSink * sink, GstPad * pad,
GstCaps * caps)
{
GstRTSPStreamContext *context;
GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad);
GstElement *payloader;
GstPad *sinkpad, *srcpad, *ghostsink;
context = gst_pad_get_element_private (pad);
if (cspad->custom_payloader) {
payloader = cspad->custom_payloader;
} else {
/* Find the payloader. */
payloader = gst_rtsp_client_sink_make_payloader (caps);
}
if (payloader == NULL)
return FALSE;
GST_DEBUG_OBJECT (sink, "Configuring payloader %" GST_PTR_FORMAT
" for pad %" GST_PTR_FORMAT, payloader, pad);
sinkpad = gst_element_get_static_pad (payloader, "sink");
if (sinkpad == NULL)
goto no_sinkpad;
srcpad = gst_element_get_static_pad (payloader, "src");
if (srcpad == NULL)
goto no_srcpad;
gst_bin_add (GST_BIN (sink->internal_bin), payloader);
ghostsink = gst_ghost_pad_new (NULL, sinkpad);
gst_pad_set_active (ghostsink, TRUE);
gst_element_add_pad (GST_ELEMENT (sink->internal_bin), ghostsink);
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_NEW_PAYLOADER], 0,
payloader);
GST_RTSP_STATE_LOCK (sink);
context->payloader_block_id =
gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM,
(GstPadProbeCallback) handle_payloader_block, context, NULL);
context->payloader = payloader;
payloader = gst_object_ref (payloader);
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), ghostsink);
gst_object_unref (GST_OBJECT (sinkpad));
GST_RTSP_STATE_UNLOCK (sink);
context->ulpfec_percentage = cspad->ulpfec_percentage;
gst_element_sync_state_with_parent (payloader);
gst_object_unref (payloader);
gst_object_unref (GST_OBJECT (srcpad));
return TRUE;
no_sinkpad:
GST_ERROR_OBJECT (sink,
"Could not find sink pad on payloader %" GST_PTR_FORMAT, payloader);
if (!cspad->custom_payloader)
gst_object_unref (payloader);
return FALSE;
no_srcpad:
GST_ERROR_OBJECT (sink,
"Could not find src pad on payloader %" GST_PTR_FORMAT, payloader);
gst_object_unref (GST_OBJECT (sinkpad));
gst_object_unref (payloader);
return TRUE;
}
static gboolean
gst_rtsp_client_sink_sinkpad_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
if (target == NULL) {
GstCaps *caps;
/* No target yet - choose a payloader and configure it */
gst_event_parse_caps (event, &caps);
GST_DEBUG_OBJECT (parent,
"Have set caps event on pad %" GST_PTR_FORMAT
" caps %" GST_PTR_FORMAT, pad, caps);
if (!gst_rtsp_client_sink_setup_payloader (GST_RTSP_CLIENT_SINK (parent),
pad, caps)) {
GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad);
GST_ELEMENT_ERROR (parent, CORE, NEGOTIATION,
("Could not create payloader"),
("Custom payloader: %p, caps: %" GST_PTR_FORMAT,
cspad->custom_payloader, caps));
gst_event_unref (event);
return FALSE;
}
} else {
gst_object_unref (target);
}
}
return gst_pad_event_default (pad, parent, event);
}
static gboolean
gst_rtsp_client_sink_sinkpad_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
if (GST_QUERY_TYPE (query) == GST_QUERY_CAPS) {
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
if (target == NULL) {
GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad);
GstCaps *caps;
if (cspad->custom_payloader) {
GstPad *sinkpad =
gst_element_get_static_pad (cspad->custom_payloader, "sink");
if (sinkpad) {
caps = gst_pad_query_caps (sinkpad, NULL);
gst_object_unref (sinkpad);
} else {
GST_ELEMENT_ERROR (parent, CORE, NEGOTIATION, (NULL),
("Custom payloaders are expected to expose a sink pad named 'sink'"));
return FALSE;
}
} else {
/* No target yet - return the union of all payloader caps */
caps = gst_rtsp_client_sink_get_all_payloaders_caps ();
}
GST_TRACE_OBJECT (parent, "Returning payloader caps %" GST_PTR_FORMAT,
caps);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
return TRUE;
}
gst_object_unref (target);
}
return gst_pad_query_default (pad, parent, query);
}
static GstPad *
gst_rtsp_client_sink_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
{
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
GstPad *pad;
GstRTSPStreamContext *context;
guint idx = (guint) - 1;
gchar *tmpname;
g_mutex_lock (&sink->preroll_lock);
if (sink->streams_collected) {
GST_WARNING_OBJECT (element, "Can't add streams to a running session");
g_mutex_unlock (&sink->preroll_lock);
return NULL;
}
g_mutex_unlock (&sink->preroll_lock);
GST_OBJECT_LOCK (sink);
if (name) {
if (!sscanf (name, "sink_%u", &idx)) {
GST_OBJECT_UNLOCK (sink);
GST_ERROR_OBJECT (element, "Invalid sink pad name %s", name);
return NULL;
}
if (idx >= sink->next_pad_id)
sink->next_pad_id = idx + 1;
}
if (idx == (guint) - 1) {
idx = sink->next_pad_id;
sink->next_pad_id++;
}
GST_OBJECT_UNLOCK (sink);
tmpname = g_strdup_printf ("sink_%u", idx);
pad = gst_rtsp_client_sink_pad_new (templ, tmpname);
g_free (tmpname);
GST_DEBUG_OBJECT (element, "Creating request pad %" GST_PTR_FORMAT, pad);
gst_pad_set_event_function (pad,
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_sinkpad_event));
gst_pad_set_query_function (pad,
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_sinkpad_query));
context = g_new0 (GstRTSPStreamContext, 1);
context->parent = sink;
context->index = idx;
gst_pad_set_element_private (pad, context);
/* The rest of the context is configured on a caps set */
gst_pad_set_active (pad, TRUE);
gst_element_add_pad (element, pad);
gst_child_proxy_child_added (GST_CHILD_PROXY (element), G_OBJECT (pad),
GST_PAD_NAME (pad));
(void) gst_rtsp_client_sink_get_factories ();
g_mutex_init (&context->conninfo.send_lock);
g_mutex_init (&context->conninfo.recv_lock);
GST_RTSP_STATE_LOCK (sink);
sink->contexts = g_list_prepend (sink->contexts, context);
GST_RTSP_STATE_UNLOCK (sink);
return pad;
}
static void
gst_rtsp_client_sink_release_pad (GstElement * element, GstPad * pad)
{
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
GstRTSPStreamContext *context;
context = gst_pad_get_element_private (pad);
/* FIXME: we may need to change our blocking state waiting for
* GstRTSPStreamBlocking messages */
GST_RTSP_STATE_LOCK (sink);
sink->contexts = g_list_remove (sink->contexts, context);
GST_RTSP_STATE_UNLOCK (sink);
/* FIXME: Shut down and clean up streaming on this pad,
* do teardown if needed */
GST_LOG_OBJECT (sink,
"Cleaning up payloader and stream for released pad %" GST_PTR_FORMAT,
pad);
if (context->stream_transport) {
gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
gst_object_unref (context->stream_transport);
context->stream_transport = NULL;
}
if (context->stream) {
if (context->joined) {
gst_rtsp_stream_leave_bin (context->stream,
GST_BIN (sink->internal_bin), sink->rtpbin);
context->joined = FALSE;
}
gst_object_unref (context->stream);
context->stream = NULL;
}
if (context->srtcpparams)
gst_caps_unref (context->srtcpparams);
g_free (context->conninfo.location);
context->conninfo.location = NULL;
g_mutex_clear (&context->conninfo.send_lock);
g_mutex_clear (&context->conninfo.recv_lock);
g_free (context);
gst_element_remove_pad (element, pad);
}
static GstClock *
gst_rtsp_client_sink_provide_clock (GstElement * element)
{
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
GstClock *clock;
if ((clock = sink->provided_clock) != NULL)
gst_object_ref (clock);
return clock;
}
/* a proxy string of the format [user:passwd@]host[:port] */
static gboolean
gst_rtsp_client_sink_set_proxy (GstRTSPClientSink * rtsp, const gchar * proxy)
{
gchar *p, *at, *col;
g_free (rtsp->proxy_user);
rtsp->proxy_user = NULL;
g_free (rtsp->proxy_passwd);
rtsp->proxy_passwd = NULL;
g_free (rtsp->proxy_host);
rtsp->proxy_host = NULL;
rtsp->proxy_port = 0;
p = (gchar *) proxy;
if (p == NULL)
return TRUE;
/* we allow http:// in front but ignore it */
if (g_str_has_prefix (p, "http://"))
p += 7;
at = strchr (p, '@');
if (at) {
/* look for user:passwd */
col = strchr (proxy, ':');
if (col == NULL || col > at)
return FALSE;
rtsp->proxy_user = g_strndup (p, col - p);
col++;
rtsp->proxy_passwd = g_strndup (col, at - col);
/* move to host */
p = at + 1;
} else {
if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
}
}
col = strchr (p, ':');
if (col) {
/* everything before the colon is the hostname */
rtsp->proxy_host = g_strndup (p, col - p);
p = col + 1;
rtsp->proxy_port = strtoul (p, (char **) &p, 10);
} else {
rtsp->proxy_host = g_strdup (p);
rtsp->proxy_port = 8080;
}
return TRUE;
}
static void
gst_rtsp_client_sink_set_tcp_timeout (GstRTSPClientSink * rtsp_client_sink,
guint64 timeout)
{
rtsp_client_sink->tcp_timeout = timeout;
}
static void
gst_rtsp_client_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTSPClientSink *rtsp_client_sink;
rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
switch (prop_id) {
case PROP_LOCATION:
gst_rtsp_client_sink_uri_set_uri (GST_URI_HANDLER (rtsp_client_sink),
g_value_get_string (value), NULL);
break;
case PROP_PROTOCOLS:
rtsp_client_sink->protocols = g_value_get_flags (value);
break;
case PROP_PROFILES:
rtsp_client_sink->profiles = g_value_get_flags (value);
break;
case PROP_DEBUG:
rtsp_client_sink->debug = g_value_get_boolean (value);
break;
case PROP_RETRY:
rtsp_client_sink->retry = g_value_get_uint (value);
break;
case PROP_TIMEOUT:
rtsp_client_sink->udp_timeout = g_value_get_uint64 (value);
break;
case PROP_TCP_TIMEOUT:
gst_rtsp_client_sink_set_tcp_timeout (rtsp_client_sink,
g_value_get_uint64 (value));
break;
case PROP_LATENCY:
rtsp_client_sink->latency = g_value_get_uint (value);
break;
case PROP_RTX_TIME:
rtsp_client_sink->rtx_time = g_value_get_uint (value);
break;
case PROP_DO_RTSP_KEEP_ALIVE:
rtsp_client_sink->do_rtsp_keep_alive = g_value_get_boolean (value);
break;
case PROP_PROXY:
gst_rtsp_client_sink_set_proxy (rtsp_client_sink,
g_value_get_string (value));
break;
case PROP_PROXY_ID:
if (rtsp_client_sink->prop_proxy_id)
g_free (rtsp_client_sink->prop_proxy_id);
rtsp_client_sink->prop_proxy_id = g_value_dup_string (value);
break;
case PROP_PROXY_PW:
if (rtsp_client_sink->prop_proxy_pw)
g_free (rtsp_client_sink->prop_proxy_pw);
rtsp_client_sink->prop_proxy_pw = g_value_dup_string (value);
break;
case PROP_RTP_BLOCKSIZE:
rtsp_client_sink->rtp_blocksize = g_value_get_uint (value);
break;
case PROP_USER_ID:
if (rtsp_client_sink->user_id)
g_free (rtsp_client_sink->user_id);
rtsp_client_sink->user_id = g_value_dup_string (value);
break;
case PROP_USER_PW:
if (rtsp_client_sink->user_pw)
g_free (rtsp_client_sink->user_pw);
rtsp_client_sink->user_pw = g_value_dup_string (value);
break;
case PROP_PORT_RANGE:
{
const gchar *str;
str = g_value_get_string (value);
if (!str || !sscanf (str, "%u-%u",
&rtsp_client_sink->client_port_range.min,
&rtsp_client_sink->client_port_range.max)) {
rtsp_client_sink->client_port_range.min = 0;
rtsp_client_sink->client_port_range.max = 0;
}
break;
}
case PROP_UDP_BUFFER_SIZE:
rtsp_client_sink->udp_buffer_size = g_value_get_int (value);
break;
case PROP_UDP_RECONNECT:
rtsp_client_sink->udp_reconnect = g_value_get_boolean (value);
break;
case PROP_MULTICAST_IFACE:
g_free (rtsp_client_sink->multi_iface);
if (g_value_get_string (value) == NULL)
rtsp_client_sink->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
else
rtsp_client_sink->multi_iface = g_value_dup_string (value);
break;
case PROP_SDES:
rtsp_client_sink->sdes = g_value_dup_boxed (value);
break;
case PROP_TLS_VALIDATION_FLAGS:
rtsp_client_sink->tls_validation_flags = g_value_get_flags (value);
break;
case PROP_TLS_DATABASE:
g_clear_object (&rtsp_client_sink->tls_database);
rtsp_client_sink->tls_database = g_value_dup_object (value);
break;
case PROP_TLS_INTERACTION:
g_clear_object (&rtsp_client_sink->tls_interaction);
rtsp_client_sink->tls_interaction = g_value_dup_object (value);
break;
case PROP_NTP_TIME_SOURCE:
rtsp_client_sink->ntp_time_source = g_value_get_enum (value);
break;
case PROP_USER_AGENT:
g_free (rtsp_client_sink->user_agent);
rtsp_client_sink->user_agent = g_value_dup_string (value);
break;
case PROP_PUBLISH_CLOCK_MODE:
rtsp_client_sink->publish_clock_mode = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtsp_client_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRTSPClientSink *rtsp_client_sink;
rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
switch (prop_id) {
case PROP_LOCATION:
g_value_set_string (value, rtsp_client_sink->conninfo.location);
break;
case PROP_PROTOCOLS:
g_value_set_flags (value, rtsp_client_sink->protocols);
break;
case PROP_PROFILES:
g_value_set_flags (value, rtsp_client_sink->profiles);
break;
case PROP_DEBUG:
g_value_set_boolean (value, rtsp_client_sink->debug);
break;
case PROP_RETRY:
g_value_set_uint (value, rtsp_client_sink->retry);
break;
case PROP_TIMEOUT:
g_value_set_uint64 (value, rtsp_client_sink->udp_timeout);
break;
case PROP_TCP_TIMEOUT:
g_value_set_uint64 (value, rtsp_client_sink->tcp_timeout);
break;
case PROP_LATENCY:
g_value_set_uint (value, rtsp_client_sink->latency);
break;
case PROP_RTX_TIME:
g_value_set_uint (value, rtsp_client_sink->rtx_time);
break;
case PROP_DO_RTSP_KEEP_ALIVE:
g_value_set_boolean (value, rtsp_client_sink->do_rtsp_keep_alive);
break;
case PROP_PROXY:
{
gchar *str;
if (rtsp_client_sink->proxy_host) {
str =
g_strdup_printf ("%s:%d", rtsp_client_sink->proxy_host,
rtsp_client_sink->proxy_port);
} else {
str = NULL;
}
g_value_take_string (value, str);
break;
}
case PROP_PROXY_ID:
g_value_set_string (value, rtsp_client_sink->prop_proxy_id);
break;
case PROP_PROXY_PW:
g_value_set_string (value, rtsp_client_sink->prop_proxy_pw);
break;
case PROP_RTP_BLOCKSIZE:
g_value_set_uint (value, rtsp_client_sink->rtp_blocksize);
break;
case PROP_USER_ID:
g_value_set_string (value, rtsp_client_sink->user_id);
break;
case PROP_USER_PW:
g_value_set_string (value, rtsp_client_sink->user_pw);
break;
case PROP_PORT_RANGE:
{
gchar *str;
if (rtsp_client_sink->client_port_range.min != 0) {
str = g_strdup_printf ("%u-%u", rtsp_client_sink->client_port_range.min,
rtsp_client_sink->client_port_range.max);
} else {
str = NULL;
}
g_value_take_string (value, str);
break;
}
case PROP_UDP_BUFFER_SIZE:
g_value_set_int (value, rtsp_client_sink->udp_buffer_size);
break;
case PROP_UDP_RECONNECT:
g_value_set_boolean (value, rtsp_client_sink->udp_reconnect);
break;
case PROP_MULTICAST_IFACE:
g_value_set_string (value, rtsp_client_sink->multi_iface);
break;
case PROP_SDES:
g_value_set_boxed (value, rtsp_client_sink->sdes);
break;
case PROP_TLS_VALIDATION_FLAGS:
g_value_set_flags (value, rtsp_client_sink->tls_validation_flags);
break;
case PROP_TLS_DATABASE:
g_value_set_object (value, rtsp_client_sink->tls_database);
break;
case PROP_TLS_INTERACTION:
g_value_set_object (value, rtsp_client_sink->tls_interaction);
break;
case PROP_NTP_TIME_SOURCE:
g_value_set_enum (value, rtsp_client_sink->ntp_time_source);
break;
case PROP_USER_AGENT:
g_value_set_string (value, rtsp_client_sink->user_agent);
break;
case PROP_PUBLISH_CLOCK_MODE:
g_value_set_enum (value, rtsp_client_sink->publish_clock_mode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static const gchar *
get_aggregate_control (GstRTSPClientSink * sink)
{
const gchar *base;
if (sink->control)
base = sink->control;
else if (sink->content_base)
base = sink->content_base;
else if (sink->conninfo.url_str)
base = sink->conninfo.url_str;
else
base = "/";
return base;
}
static void
gst_rtsp_client_sink_cleanup (GstRTSPClientSink * sink)
{
GList *walk;
GST_DEBUG_OBJECT (sink, "cleanup");
gst_element_set_state (GST_ELEMENT (sink->internal_bin), GST_STATE_NULL);
/* Clean up any left over stream objects */
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) (walk->data);
if (context->stream_transport) {
gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
gst_object_unref (context->stream_transport);
context->stream_transport = NULL;
}
if (context->stream) {
if (context->joined) {
gst_rtsp_stream_leave_bin (context->stream,
GST_BIN (sink->internal_bin), sink->rtpbin);
context->joined = FALSE;
}
gst_object_unref (context->stream);
context->stream = NULL;
}
if (context->srtcpparams) {
gst_caps_unref (context->srtcpparams);
context->srtcpparams = NULL;
}
g_free (context->conninfo.location);
context->conninfo.location = NULL;
}
if (sink->rtpbin) {
gst_element_set_state (sink->rtpbin, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (sink->internal_bin), sink->rtpbin);
sink->rtpbin = NULL;
}
g_free (sink->content_base);
sink->content_base = NULL;
g_free (sink->control);
sink->control = NULL;
if (sink->range)
gst_rtsp_range_free (sink->range);
sink->range = NULL;
/* don't clear the SDP when it was used in the url */
if (sink->uri_sdp && !sink->from_sdp) {
gst_sdp_message_free (sink->uri_sdp);
sink->uri_sdp = NULL;
}
if (sink->provided_clock) {
gst_object_unref (sink->provided_clock);
sink->provided_clock = NULL;
}
g_free (sink->server_ip);
sink->server_ip = NULL;
sink->next_pad_id = 0;
sink->next_dyn_pt = 96;
}
static GstRTSPResult
gst_rtsp_client_sink_connection_send (GstRTSPClientSink * sink,
GstRTSPConnInfo * conninfo, GstRTSPMessage * message, gint64 timeout)
{
GstRTSPResult ret;
if (conninfo->connection) {
g_mutex_lock (&conninfo->send_lock);
ret =
gst_rtsp_connection_send_usec (conninfo->connection, message, timeout);
g_mutex_unlock (&conninfo->send_lock);
} else {
ret = GST_RTSP_ERROR;
}
return ret;
}
static GstRTSPResult
gst_rtsp_client_sink_connection_send_messages (GstRTSPClientSink * sink,
GstRTSPConnInfo * conninfo, GstRTSPMessage * messages, guint n_messages,
gint64 timeout)
{
GstRTSPResult ret;
if (conninfo->connection) {
g_mutex_lock (&conninfo->send_lock);
ret =
gst_rtsp_connection_send_messages_usec (conninfo->connection, messages,
n_messages, timeout);
g_mutex_unlock (&conninfo->send_lock);
} else {
ret = GST_RTSP_ERROR;
}
return ret;
}
static GstRTSPResult
gst_rtsp_client_sink_connection_receive (GstRTSPClientSink * sink,
GstRTSPConnInfo * conninfo, GstRTSPMessage * message, gint64 timeout)
{
GstRTSPResult ret;
if (conninfo->connection) {
g_mutex_lock (&conninfo->recv_lock);
ret = gst_rtsp_connection_receive_usec (conninfo->connection, message,
timeout);
g_mutex_unlock (&conninfo->recv_lock);
} else {
ret = GST_RTSP_ERROR;
}
return ret;
}
static gboolean
accept_certificate_cb (GTlsConnection * conn, GTlsCertificate * peer_cert,
GTlsCertificateFlags errors, gpointer user_data)
{
GstRTSPClientSink *sink = user_data;
gboolean accept = FALSE;
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_ACCEPT_CERTIFICATE],
0, conn, peer_cert, errors, &accept);
return accept;
}
static GstRTSPResult
gst_rtsp_conninfo_connect (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
gboolean async)
{
GstRTSPResult res;
if (info->connection == NULL) {
if (info->url == NULL) {
GST_DEBUG_OBJECT (sink, "parsing uri (%s)...", info->location);
if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
goto parse_error;
}
/* create connection */
GST_DEBUG_OBJECT (sink, "creating connection (%s)...", info->location);
if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
goto could_not_create;
if (info->url_str)
g_free (info->url_str);
info->url_str = gst_rtsp_url_get_request_uri (info->url);
GST_DEBUG_OBJECT (sink, "sanitized uri %s", info->url_str);
if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
sink->tls_validation_flags))
GST_WARNING_OBJECT (sink, "Unable to set TLS validation flags");
if (sink->tls_database)
gst_rtsp_connection_set_tls_database (info->connection,
sink->tls_database);
if (sink->tls_interaction)
gst_rtsp_connection_set_tls_interaction (info->connection,
sink->tls_interaction);
gst_rtsp_connection_set_accept_certificate_func (info->connection,
accept_certificate_cb, sink, NULL);
}
if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
gst_rtsp_connection_set_tunneled (info->connection, TRUE);
if (sink->proxy_host) {
GST_DEBUG_OBJECT (sink, "setting proxy %s:%d", sink->proxy_host,
sink->proxy_port);
gst_rtsp_connection_set_proxy (info->connection, sink->proxy_host,
sink->proxy_port);
}
}
if (!info->connected) {
/* connect */
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "connect",
("Connecting to %s", info->location));
GST_DEBUG_OBJECT (sink, "connecting (%s)...", info->location);
if ((res =
gst_rtsp_connection_connect_usec (info->connection,
sink->tcp_timeout)) < 0)
goto could_not_connect;
info->connected = TRUE;
}
return GST_RTSP_OK;
/* ERRORS */
parse_error:
{
GST_ERROR_OBJECT (sink, "No valid RTSP URL was provided");
return res;
}
could_not_create:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR_OBJECT (sink, "Could not create connection. (%s)", str);
g_free (str);
return res;
}
could_not_connect:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR_OBJECT (sink, "Could not connect to server. (%s)", str);
g_free (str);
return res;
}
}
static GstRTSPResult
gst_rtsp_conninfo_close (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
gboolean free)
{
GST_RTSP_STATE_LOCK (sink);
if (info->connected) {
GST_DEBUG_OBJECT (sink, "closing connection...");
gst_rtsp_connection_close (info->connection);
info->connected = FALSE;
}
if (free && info->connection) {
/* free connection */
GST_DEBUG_OBJECT (sink, "freeing connection...");
gst_rtsp_connection_free (info->connection);
g_mutex_lock (&sink->preroll_lock);
info->connection = NULL;
g_cond_broadcast (&sink->preroll_cond);
g_mutex_unlock (&sink->preroll_lock);
}
GST_RTSP_STATE_UNLOCK (sink);
return GST_RTSP_OK;
}
static GstRTSPResult
gst_rtsp_conninfo_reconnect (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
gboolean async)
{
GstRTSPResult res;
GST_DEBUG_OBJECT (sink, "reconnecting connection...");
gst_rtsp_conninfo_close (sink, info, FALSE);
res = gst_rtsp_conninfo_connect (sink, info, async);
return res;
}
static void
gst_rtsp_client_sink_connection_flush (GstRTSPClientSink * sink, gboolean flush)
{
GList *walk;
GST_DEBUG_OBJECT (sink, "set flushing %d", flush);
g_mutex_lock (&sink->preroll_lock);
if (sink->conninfo.connection && sink->conninfo.flushing != flush) {
GST_DEBUG_OBJECT (sink, "connection flush");
gst_rtsp_connection_flush (sink->conninfo.connection, flush);
sink->conninfo.flushing = flush;
}
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
GST_DEBUG_OBJECT (sink, "stream %p flush", stream);
gst_rtsp_connection_flush (stream->conninfo.connection, flush);
stream->conninfo.flushing = flush;
}
}
g_cond_broadcast (&sink->preroll_cond);
g_mutex_unlock (&sink->preroll_lock);
}
static GstRTSPResult
gst_rtsp_client_sink_init_request (GstRTSPClientSink * sink,
GstRTSPMessage * msg, GstRTSPMethod method, const gchar * uri)
{
GstRTSPResult res;
res = gst_rtsp_message_init_request (msg, method, uri);
if (res < 0)
return res;
/* set user-agent */
if (sink->user_agent)
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_USER_AGENT,
sink->user_agent);
return res;
}
/* FIXME, handle server request, reply with OK, for now */
static GstRTSPResult
gst_rtsp_client_sink_handle_request (GstRTSPClientSink * sink,
GstRTSPConnInfo * conninfo, GstRTSPMessage * request)
{
GstRTSPMessage response = { 0 };
GstRTSPResult res;
GST_DEBUG_OBJECT (sink, "got server request message");
if (sink->debug)
gst_rtsp_message_dump (request);
/* default implementation, send OK */
GST_DEBUG_OBJECT (sink, "prepare OK reply");
res =
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
request);
if (res < 0)
goto send_error;
/* let app parse and reply */
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_HANDLE_REQUEST],
0, request, &response);
if (sink->debug)
gst_rtsp_message_dump (&response);
res = gst_rtsp_client_sink_connection_send (sink, conninfo, &response, 0);
if (res < 0)
goto send_error;
gst_rtsp_message_unset (&response);
return GST_RTSP_OK;
/* ERRORS */
send_error:
{
gst_rtsp_message_unset (&response);
return res;
}
}
/* send server keep-alive */
static GstRTSPResult
gst_rtsp_client_sink_send_keep_alive (GstRTSPClientSink * sink)
{
GstRTSPMessage request = { 0 };
GstRTSPResult res;
GstRTSPMethod method;
const gchar *control;
if (sink->do_rtsp_keep_alive == FALSE) {
GST_DEBUG_OBJECT (sink, "do-rtsp-keep-alive is FALSE, not sending.");
gst_rtsp_connection_reset_timeout (sink->conninfo.connection);
return GST_RTSP_OK;
}
GST_DEBUG_OBJECT (sink, "creating server keep-alive");
/* find a method to use for keep-alive */
if (sink->methods & GST_RTSP_GET_PARAMETER)
method = GST_RTSP_GET_PARAMETER;
else
method = GST_RTSP_OPTIONS;
control = get_aggregate_control (sink);
if (control == NULL)
goto no_control;
res = gst_rtsp_client_sink_init_request (sink, &request, method, control);
if (res < 0)
goto send_error;
if (sink->debug)
gst_rtsp_message_dump (&request);
res =
gst_rtsp_client_sink_connection_send (sink, &sink->conninfo, &request, 0);
if (res < 0)
goto send_error;
gst_rtsp_connection_reset_timeout (sink->conninfo.connection);
gst_rtsp_message_unset (&request);
return GST_RTSP_OK;
/* ERRORS */
no_control:
{
GST_WARNING_OBJECT (sink, "no control url to send keepalive");
return GST_RTSP_OK;
}
send_error:
{
gchar *str = gst_rtsp_strresult (res);
gst_rtsp_message_unset (&request);
GST_ELEMENT_WARNING (sink, RESOURCE, WRITE, (NULL),
("Could not send keep-alive. (%s)", str));
g_free (str);
return res;
}
}
static GstFlowReturn
gst_rtsp_client_sink_loop_rx (GstRTSPClientSink * sink)
{
GstRTSPResult res;
GstRTSPMessage message = { 0 };
gint retry = 0;
while (TRUE) {
gint64 timeout;
/* get the next timeout interval */
timeout = gst_rtsp_connection_next_timeout_usec (sink->conninfo.connection);
GST_DEBUG_OBJECT (sink, "doing receive with timeout %d seconds",
(gint) timeout / G_USEC_PER_SEC);
gst_rtsp_message_unset (&message);
/* we should continue reading the TCP socket because the server might
* send us requests. When the session timeout expires, we need to send a
* keep-alive request to keep the session open. */
res =
gst_rtsp_client_sink_connection_receive (sink,
&sink->conninfo, &message, timeout);
switch (res) {
case GST_RTSP_OK:
GST_DEBUG_OBJECT (sink, "we received a server message");
break;
case GST_RTSP_EINTR:
/* we got interrupted, see what we have to do */
goto interrupt;
case GST_RTSP_ETIMEOUT:
/* send keep-alive, ignore the result, a warning will be posted. */
GST_DEBUG_OBJECT (sink, "timeout, sending keep-alive");
if ((res =
gst_rtsp_client_sink_send_keep_alive (sink)) == GST_RTSP_EINTR)
goto interrupt;
continue;
case GST_RTSP_EEOF:
/* server closed the connection. not very fatal for UDP, reconnect and
* see what happens. */
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
("The server closed the connection."));
if (sink->udp_reconnect) {
if ((res =
gst_rtsp_conninfo_reconnect (sink, &sink->conninfo,
FALSE)) < 0)
goto connect_error;
} else {
goto server_eof;
}
continue;
break;
case GST_RTSP_ENET:
GST_DEBUG_OBJECT (sink, "An ethernet problem occured.");
default:
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
("Unhandled return value %d.", res));
goto receive_error;
}
switch (message.type) {
case GST_RTSP_MESSAGE_REQUEST:
/* server sends us a request message, handle it */
res =
gst_rtsp_client_sink_handle_request (sink,
&sink->conninfo, &message);
if (res == GST_RTSP_EEOF)
goto server_eof;
else if (res < 0)
goto handle_request_failed;
break;
case GST_RTSP_MESSAGE_RESPONSE:
/* we ignore response and data messages */
GST_DEBUG_OBJECT (sink, "ignoring response message");
if (sink->debug)
gst_rtsp_message_dump (&message);
if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
GST_DEBUG_OBJECT (sink, "but is Unauthorized response ...");
if (gst_rtsp_client_sink_setup_auth (sink, &message) && !(retry++)) {
GST_DEBUG_OBJECT (sink, "so retrying keep-alive");
if ((res =
gst_rtsp_client_sink_send_keep_alive (sink)) ==
GST_RTSP_EINTR)
goto interrupt;
}
} else {
retry = 0;
}
break;
case GST_RTSP_MESSAGE_DATA:
/* we ignore response and data messages */
GST_DEBUG_OBJECT (sink, "ignoring data message");
break;
default:
GST_WARNING_OBJECT (sink, "ignoring unknown message type %d",
message.type);
break;
}
}
g_assert_not_reached ();
/* we get here when the connection got interrupted */
interrupt:
{
gst_rtsp_message_unset (&message);
GST_DEBUG_OBJECT (sink, "got interrupted");
return GST_FLOW_FLUSHING;
}
connect_error:
{
gchar *str = gst_rtsp_strresult (res);
GstFlowReturn ret;
sink->conninfo.connected = FALSE;
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ_WRITE, (NULL),
("Could not connect to server. (%s)", str));
g_free (str);
ret = GST_FLOW_ERROR;
} else {
ret = GST_FLOW_FLUSHING;
}
return ret;
}
receive_error:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
g_free (str);
return GST_FLOW_ERROR;
}
handle_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GstFlowReturn ret;
gst_rtsp_message_unset (&message);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Could not handle server message. (%s)", str));
g_free (str);
ret = GST_FLOW_ERROR;
} else {
ret = GST_FLOW_FLUSHING;
}
return ret;
}
server_eof:
{
GST_DEBUG_OBJECT (sink, "we got an eof from the server");
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
("The server closed the connection."));
sink->conninfo.connected = FALSE;
gst_rtsp_message_unset (&message);
return GST_FLOW_EOS;
}
}
static GstRTSPResult
gst_rtsp_client_sink_reconnect (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPResult res = GST_RTSP_OK;
gboolean restart = FALSE;
GST_DEBUG_OBJECT (sink, "doing reconnect");
GST_FIXME_OBJECT (sink, "Reconnection is not yet implemented");
/* no need to restart, we're done */
if (!restart)
goto done;
/* we can try only TCP now */
sink->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
/* close and cleanup our state */
if ((res = gst_rtsp_client_sink_close (sink, async, FALSE)) < 0)
goto done;
/* see if we have TCP left to try. Also don't try TCP when we were configured
* with an SDP. */
if (!(sink->protocols & GST_RTSP_LOWER_TRANS_TCP) || sink->from_sdp)
goto no_protocols;
/* We post a warning message now to inform the user
* that nothing happened. It's most likely a firewall thing. */
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
"firewall is blocking it. Retrying using a TCP connection.",
gst_guint64_to_gdouble (sink->udp_timeout / 1000000.0)));
/* open new connection using tcp */
if (gst_rtsp_client_sink_open (sink, async) < 0)
goto open_failed;
/* start recording */
if (gst_rtsp_client_sink_record (sink, async) < 0)
goto play_failed;
done:
return res;
/* ERRORS */
no_protocols:
{
sink->cur_protocols = 0;
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
"firewall is blocking it. No other protocols to try.",
gst_guint64_to_gdouble (sink->udp_timeout / 1000000.0)));
return GST_RTSP_ERROR;
}
open_failed:
{
GST_DEBUG_OBJECT (sink, "open failed");
return GST_RTSP_OK;
}
play_failed:
{
GST_DEBUG_OBJECT (sink, "play failed");
return GST_RTSP_OK;
}
}
static void
gst_rtsp_client_sink_loop_start_cmd (GstRTSPClientSink * sink, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (sink, START, "open", ("Opening Stream"));
break;
case CMD_RECORD:
GST_ELEMENT_PROGRESS (sink, START, "request", ("Sending RECORD request"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (sink, START, "request", ("Sending PAUSE request"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (sink, START, "close", ("Closing Stream"));
break;
default:
break;
}
}
static void
gst_rtsp_client_sink_loop_complete_cmd (GstRTSPClientSink * sink, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (sink, COMPLETE, "open", ("Opened Stream"));
break;
case CMD_RECORD:
GST_ELEMENT_PROGRESS (sink, COMPLETE, "request", ("Sent RECORD request"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (sink, COMPLETE, "request", ("Sent PAUSE request"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (sink, COMPLETE, "close", ("Closed Stream"));
break;
default:
break;
}
}
static void
gst_rtsp_client_sink_loop_cancel_cmd (GstRTSPClientSink * sink, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (sink, CANCELED, "open", ("Open canceled"));
break;
case CMD_RECORD:
GST_ELEMENT_PROGRESS (sink, CANCELED, "request", ("RECORD canceled"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (sink, CANCELED, "request", ("PAUSE canceled"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (sink, CANCELED, "close", ("Close canceled"));
break;
default:
break;
}
}
static void
gst_rtsp_client_sink_loop_error_cmd (GstRTSPClientSink * sink, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (sink, ERROR, "open", ("Open failed"));
break;
case CMD_RECORD:
GST_ELEMENT_PROGRESS (sink, ERROR, "request", ("RECORD failed"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (sink, ERROR, "request", ("PAUSE failed"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (sink, ERROR, "close", ("Close failed"));
break;
default:
break;
}
}
static void
gst_rtsp_client_sink_loop_end_cmd (GstRTSPClientSink * sink, gint cmd,
GstRTSPResult ret)
{
if (ret == GST_RTSP_OK)
gst_rtsp_client_sink_loop_complete_cmd (sink, cmd);
else if (ret == GST_RTSP_EINTR)
gst_rtsp_client_sink_loop_cancel_cmd (sink, cmd);
else
gst_rtsp_client_sink_loop_error_cmd (sink, cmd);
}
static gboolean
gst_rtsp_client_sink_loop_send_cmd (GstRTSPClientSink * sink, gint cmd,
gint mask)
{
gint old;
gboolean flushed = FALSE;
/* start new request */
gst_rtsp_client_sink_loop_start_cmd (sink, cmd);
GST_DEBUG_OBJECT (sink, "sending cmd %s", cmd_to_string (cmd));
GST_OBJECT_LOCK (sink);
old = sink->pending_cmd;
if (old == CMD_RECONNECT) {
GST_DEBUG_OBJECT (sink, "ignore, we were reconnecting");
cmd = CMD_RECONNECT;
}
if (old != CMD_WAIT) {
sink->pending_cmd = CMD_WAIT;
GST_OBJECT_UNLOCK (sink);
/* cancel previous request */
GST_DEBUG_OBJECT (sink, "cancel previous request %s", cmd_to_string (old));
gst_rtsp_client_sink_loop_cancel_cmd (sink, old);
GST_OBJECT_LOCK (sink);
}
sink->pending_cmd = cmd;
/* interrupt if allowed */
if (sink->busy_cmd & mask) {
GST_DEBUG_OBJECT (sink, "connection flush busy %s",
cmd_to_string (sink->busy_cmd));
gst_rtsp_client_sink_connection_flush (sink, TRUE);
flushed = TRUE;
} else {
GST_DEBUG_OBJECT (sink, "not interrupting busy cmd %s",
cmd_to_string (sink->busy_cmd));
}
if (sink->task)
gst_task_start (sink->task);
GST_OBJECT_UNLOCK (sink);
return flushed;
}
static gboolean
gst_rtsp_client_sink_loop (GstRTSPClientSink * sink)
{
GstFlowReturn ret;
if (!sink->conninfo.connection || !sink->conninfo.connected)
goto no_connection;
ret = gst_rtsp_client_sink_loop_rx (sink);
if (ret != GST_FLOW_OK)
goto pause;
return TRUE;
/* ERRORS */
no_connection:
{
GST_WARNING_OBJECT (sink, "we are not connected");
ret = GST_FLOW_FLUSHING;
goto pause;
}
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (sink, "pausing task, reason %s", reason);
gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_LOOP);
return FALSE;
}
}
#ifndef GST_DISABLE_GST_DEBUG
static const gchar *
gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
{
gint index = 0;
while (method != 0) {
index++;
method >>= 1;
}
switch (index) {
case 0:
return "None";
case 1:
return "Basic";
case 2:
return "Digest";
}
return "Unknown";
}
#endif
/* Parse a WWW-Authenticate Response header and determine the
* available authentication methods
*
* This code should also cope with the fact that each WWW-Authenticate
* header can contain multiple challenge methods + tokens
*
* At the moment, for Basic auth, we just do a minimal check and don't
* even parse out the realm */
static void
gst_rtsp_client_sink_parse_auth_hdr (GstRTSPMessage * response,
GstRTSPAuthMethod * methods, GstRTSPConnection * conn, gboolean * stale)
{
GstRTSPAuthCredential **credentials, **credential;
g_return_if_fail (response != NULL);
g_return_if_fail (methods != NULL);
g_return_if_fail (stale != NULL);
credentials =
gst_rtsp_message_parse_auth_credentials (response,
GST_RTSP_HDR_WWW_AUTHENTICATE);
if (!credentials)
return;
credential = credentials;
while (*credential) {
if ((*credential)->scheme == GST_RTSP_AUTH_BASIC) {
*methods |= GST_RTSP_AUTH_BASIC;
} else if ((*credential)->scheme == GST_RTSP_AUTH_DIGEST) {
GstRTSPAuthParam **param = (*credential)->params;
*methods |= GST_RTSP_AUTH_DIGEST;
gst_rtsp_connection_clear_auth_params (conn);
*stale = FALSE;
while (*param) {
if (strcmp ((*param)->name, "stale") == 0
&& g_ascii_strcasecmp ((*param)->value, "TRUE") == 0)
*stale = TRUE;
gst_rtsp_connection_set_auth_param (conn, (*param)->name,
(*param)->value);
param++;
}
}
credential++;
}
gst_rtsp_auth_credentials_free (credentials);
}
/**
* gst_rtsp_client_sink_setup_auth:
* @src: the rtsp source
*
* Configure a username and password and auth method on the
* connection object based on a response we received from the
* peer.
*
* Currently, this requires that a username and password were supplied
* in the uri. In the future, they may be requested on demand by sending
* a message up the bus.
*
* Returns: TRUE if authentication information could be set up correctly.
*/
static gboolean
gst_rtsp_client_sink_setup_auth (GstRTSPClientSink * sink,
GstRTSPMessage * response)
{
gchar *user = NULL;
gchar *pass = NULL;
GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
GstRTSPAuthMethod method;
GstRTSPResult auth_result;
GstRTSPUrl *url;
GstRTSPConnection *conn;
gboolean stale = FALSE;
conn = sink->conninfo.connection;
/* Identify the available auth methods and see if any are supported */
gst_rtsp_client_sink_parse_auth_hdr (response, &avail_methods, conn, &stale);
if (avail_methods == GST_RTSP_AUTH_NONE)
goto no_auth_available;
/* For digest auth, if the response indicates that the session
* data are stale, we just update them in the connection object and
* return TRUE to retry the request */
if (stale)
sink->tried_url_auth = FALSE;
url = gst_rtsp_connection_get_url (conn);
/* Do we have username and password available? */
if (url != NULL && !sink->tried_url_auth && url->user != NULL
&& url->passwd != NULL) {
user = url->user;
pass = url->passwd;
sink->tried_url_auth = TRUE;
GST_DEBUG_OBJECT (sink,
"Attempting authentication using credentials from the URL");
} else {
user = sink->user_id;
pass = sink->user_pw;
GST_DEBUG_OBJECT (sink,
"Attempting authentication using credentials from the properties");
}
/* FIXME: If the url didn't contain username and password or we tried them
* already, request a username and passwd from the application via some kind
* of credentials request message */
/* If we don't have a username and passwd at this point, bail out. */
if (user == NULL || pass == NULL)
goto no_user_pass;
/* Try to configure for each available authentication method, strongest to
* weakest */
for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
/* Check if this method is available on the server */
if ((method & avail_methods) == 0)
continue;
/* Pass the credentials to the connection to try on the next request */
auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
/* INVAL indicates an invalid username/passwd were supplied, so we'll just
* ignore it and end up retrying later */
if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
GST_DEBUG_OBJECT (sink, "Attempting %s authentication",
gst_rtsp_auth_method_to_string (method));
break;
}
}
if (method == GST_RTSP_AUTH_NONE)
goto no_auth_available;
return TRUE;
no_auth_available:
{
/* Output an error indicating that we couldn't connect because there were
* no supported authentication protocols */
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ, (NULL),
("No supported authentication protocol was found"));
return FALSE;
}
no_user_pass:
{
/* We don't fire an error message, we just return FALSE and let the
* normal NOT_AUTHORIZED error be propagated */
return FALSE;
}
}
static GstRTSPResult
gst_rtsp_client_sink_try_send (GstRTSPClientSink * sink,
GstRTSPConnInfo * conninfo, GstRTSPMessage * requests,
guint n_requests, GstRTSPMessage * response, GstRTSPStatusCode * code)
{
GstRTSPResult res;
GstRTSPStatusCode thecode;
gchar *content_base = NULL;
gint try = 0;
g_assert (n_requests == 1 || response == NULL);
again:
GST_DEBUG_OBJECT (sink, "sending message");
if (sink->debug && n_requests == 1)
gst_rtsp_message_dump (&requests[0]);
g_mutex_lock (&sink->send_lock);
res =
gst_rtsp_client_sink_connection_send_messages (sink, conninfo, requests,
n_requests, sink->tcp_timeout);
if (res < 0) {
g_mutex_unlock (&sink->send_lock);
goto send_error;
}
gst_rtsp_connection_reset_timeout (conninfo->connection);
/* See if we should handle the response */
if (response == NULL) {
g_mutex_unlock (&sink->send_lock);
return GST_RTSP_OK;
}
next:
res =
gst_rtsp_client_sink_connection_receive (sink, conninfo, response,
sink->tcp_timeout);
g_mutex_unlock (&sink->send_lock);
if (res < 0)
goto receive_error;
if (sink->debug)
gst_rtsp_message_dump (response);
switch (response->type) {
case GST_RTSP_MESSAGE_REQUEST:
res = gst_rtsp_client_sink_handle_request (sink, conninfo, response);
if (res == GST_RTSP_EEOF)
goto server_eof;
else if (res < 0)
goto handle_request_failed;
g_mutex_lock (&sink->send_lock);
goto next;
case GST_RTSP_MESSAGE_RESPONSE:
/* ok, a response is good */
GST_DEBUG_OBJECT (sink, "received response message");
break;
case GST_RTSP_MESSAGE_DATA:
/* we ignore data messages */
GST_DEBUG_OBJECT (sink, "ignoring data message");
g_mutex_lock (&sink->send_lock);
goto next;
default:
GST_WARNING_OBJECT (sink, "ignoring unknown message type %d",
response->type);
g_mutex_lock (&sink->send_lock);
goto next;
}
thecode = response->type_data.response.code;
GST_DEBUG_OBJECT (sink, "got response message %d", thecode);
/* if the caller wanted the result code, we store it. */
if (code)
*code = thecode;
/* If the request didn't succeed, bail out before doing any more */
if (thecode != GST_RTSP_STS_OK)
return GST_RTSP_OK;
/* store new content base if any */
gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
&content_base, 0);
if (content_base) {
g_free (sink->content_base);
sink->content_base = g_strdup (content_base);
}
return GST_RTSP_OK;
/* ERRORS */
send_error:
{
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "send interrupted");
}
g_free (str);
return res;
}
receive_error:
{
switch (res) {
case GST_RTSP_EEOF:
GST_WARNING_OBJECT (sink, "server closed connection");
if ((try == 0) && !sink->interleaved && sink->udp_reconnect) {
try++;
/* if reconnect succeeds, try again */
if ((res =
gst_rtsp_conninfo_reconnect (sink, &sink->conninfo,
FALSE)) == 0)
goto again;
}
/* only try once after reconnect, then fallthrough and error out */
default:
{
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "receive interrupted");
}
g_free (str);
break;
}
}
return res;
}
handle_request_failed:
{
/* ERROR was posted */
gst_rtsp_message_unset (response);
return res;
}
server_eof:
{
GST_DEBUG_OBJECT (sink, "we got an eof from the server");
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
("The server closed the connection."));
gst_rtsp_message_unset (response);
return res;
}
}
static void
gst_rtsp_client_sink_set_state (GstRTSPClientSink * sink, GstState state)
{
GST_DEBUG_OBJECT (sink, "Setting internal state to %s",
gst_element_state_get_name (state));
gst_element_set_state (GST_ELEMENT (sink->internal_bin), state);
}
/**
* gst_rtsp_client_sink_send:
* @src: the rtsp source
* @conn: the connection to send on
* @request: must point to a valid request
* @response: must point to an empty #GstRTSPMessage
* @code: an optional code result
*
* send @request and retrieve the response in @response. optionally @code can be
* non-NULL in which case it will contain the status code of the response.
*
* If This function returns #GST_RTSP_OK, @response will contain a valid response
* message that should be cleaned with gst_rtsp_message_unset() after usage.
*
* If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
* @response message) if the response code was not 200 (OK).
*
* If the attempt results in an authentication failure, then this will attempt
* to retrieve authentication credentials via gst_rtsp_client_sink_setup_auth and retry
* the request.
*
* Returns: #GST_RTSP_OK if the processing was successful.
*/
static GstRTSPResult
gst_rtsp_client_sink_send (GstRTSPClientSink * sink, GstRTSPConnInfo * conninfo,
GstRTSPMessage * request, GstRTSPMessage * response,
GstRTSPStatusCode * code)
{
GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
GstRTSPResult res = GST_RTSP_ERROR;
gint count;
gboolean retry;
GstRTSPMethod method = GST_RTSP_INVALID;
count = 0;
do {
retry = FALSE;
/* make sure we don't loop forever */
if (count++ > 8)
break;
/* save method so we can disable it when the server complains */
method = request->type_data.request.method;
if ((res =
gst_rtsp_client_sink_try_send (sink, conninfo, request, 1, response,
&int_code)) < 0)
goto error;
switch (int_code) {
case GST_RTSP_STS_UNAUTHORIZED:
if (gst_rtsp_client_sink_setup_auth (sink, response)) {
/* Try the request/response again after configuring the auth info
* and loop again */
retry = TRUE;
}
break;
default:
break;
}
} while (retry == TRUE);
/* If the user requested the code, let them handle errors, otherwise
* post an error below */
if (code != NULL)
*code = int_code;
else if (int_code != GST_RTSP_STS_OK)
goto error_response;
return res;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (sink, "got error %d", res);
return res;
}
error_response:
{
res = GST_RTSP_ERROR;
switch (response->type_data.response.code) {
case GST_RTSP_STS_NOT_FOUND:
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, (NULL), ("%s",
response->type_data.response.reason));
break;
case GST_RTSP_STS_UNAUTHORIZED:
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_AUTHORIZED, (NULL), ("%s",
response->type_data.response.reason));
break;
case GST_RTSP_STS_MOVED_PERMANENTLY:
case GST_RTSP_STS_MOVE_TEMPORARILY:
{
gchar *new_location;
GstRTSPLowerTrans transports;
GST_DEBUG_OBJECT (sink, "got redirection");
/* if we don't have a Location Header, we must error */
if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
&new_location, 0) < 0)
break;
/* When we receive a redirect result, we go back to the INIT state after
* parsing the new URI. The caller should do the needed steps to issue
* a new setup when it detects this state change. */
GST_DEBUG_OBJECT (sink, "redirection to %s", new_location);
/* save current transports */
if (sink->conninfo.url)
transports = sink->conninfo.url->transports;
else
transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
gst_rtsp_client_sink_uri_set_uri (GST_URI_HANDLER (sink), new_location,
NULL);
/* set old transports */
if (sink->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
sink->conninfo.url->transports = transports;
sink->need_redirect = TRUE;
sink->state = GST_RTSP_STATE_INIT;
res = GST_RTSP_OK;
break;
}
case GST_RTSP_STS_NOT_ACCEPTABLE:
case GST_RTSP_STS_NOT_IMPLEMENTED:
case GST_RTSP_STS_METHOD_NOT_ALLOWED:
GST_WARNING_OBJECT (sink, "got NOT IMPLEMENTED, disable method %s",
gst_rtsp_method_as_text (method));
sink->methods &= ~method;
res = GST_RTSP_OK;
break;
default:
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Got error response: %d (%s).", response->type_data.response.code,
response->type_data.response.reason));
break;
}
/* if we return ERROR we should unset the response ourselves */
if (res == GST_RTSP_ERROR)
gst_rtsp_message_unset (response);
return res;
}
}
/* parse the response and collect all the supported methods. We need this
* information so that we don't try to send an unsupported request to the
* server.
*/
static gboolean
gst_rtsp_client_sink_parse_methods (GstRTSPClientSink * sink,
GstRTSPMessage * response)
{
GstRTSPHeaderField field;
gchar *respoptions;
gint indx = 0;
/* reset supported methods */
sink->methods = 0;
/* Try Allow Header first */
field = GST_RTSP_HDR_ALLOW;
while (TRUE) {
respoptions = NULL;
gst_rtsp_message_get_header (response, field, &respoptions, indx);
if (indx == 0 && !respoptions) {
/* if no Allow header was found then try the Public header... */
field = GST_RTSP_HDR_PUBLIC;
gst_rtsp_message_get_header (response, field, &respoptions, indx);
}
if (!respoptions)
break;
sink->methods |= gst_rtsp_options_from_text (respoptions);
indx++;
}
if (sink->methods == 0) {
/* neither Allow nor Public are required, assume the server supports
* at least SETUP. */
GST_DEBUG_OBJECT (sink, "could not get OPTIONS");
sink->methods = GST_RTSP_SETUP;
}
/* Even if the server replied, and didn't say it supports
* RECORD|ANNOUNCE, try anyway by assuming it does */
sink->methods |= GST_RTSP_ANNOUNCE | GST_RTSP_RECORD;
if (!(sink->methods & GST_RTSP_SETUP))
goto no_setup;
return TRUE;
/* ERRORS */
no_setup:
{
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ, (NULL),
("Server does not support SETUP."));
return FALSE;
}
}
static GstRTSPResult
gst_rtsp_client_sink_connect_to_server (GstRTSPClientSink * sink,
gboolean async)
{
GstRTSPResult res;
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GSocket *conn_socket;
GSocketAddress *sa;
GInetAddress *ia;
sink->need_redirect = FALSE;
/* can't continue without a valid url */
if (G_UNLIKELY (sink->conninfo.url == NULL)) {
res = GST_RTSP_EINVAL;
goto no_url;
}
sink->tried_url_auth = FALSE;
if ((res = gst_rtsp_conninfo_connect (sink, &sink->conninfo, async)) < 0)
goto connect_failed;
conn_socket = gst_rtsp_connection_get_read_socket (sink->conninfo.connection);
sa = g_socket_get_remote_address (conn_socket, NULL);
ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
g_free (sink->server_ip);
sink->server_ip = g_inet_address_to_string (ia);
g_object_unref (sa);
/* create OPTIONS */
GST_DEBUG_OBJECT (sink, "create options...");
res =
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_OPTIONS,
sink->conninfo.url_str);
if (res < 0)
goto create_request_failed;
/* send OPTIONS */
GST_DEBUG_OBJECT (sink, "send options...");
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "open",
("Retrieving server options"));
if ((res =
gst_rtsp_client_sink_send (sink, &sink->conninfo, &request,
&response, NULL)) < 0)
goto send_error;
/* parse OPTIONS */
if (!gst_rtsp_client_sink_parse_methods (sink, &response))
goto methods_error;
/* FIXME: Do we need to handle REDIRECT responses for OPTIONS? */
/* clean up any messages */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
/* ERRORS */
no_url:
{
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, (NULL),
("No valid RTSP URL was provided"));
goto cleanup_error;
}
connect_failed:
{
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ_WRITE, (NULL),
("Failed to connect. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "connect interrupted");
}
g_free (str);
goto cleanup_error;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto cleanup_error;
}
send_error:
{
/* Don't post a message - the rtsp_send method will have
* taken care of it because we passed NULL for the response code */
goto cleanup_error;
}
methods_error:
{
/* error was posted */
res = GST_RTSP_ERROR;
goto cleanup_error;
}
cleanup_error:
{
if (sink->conninfo.connection) {
GST_DEBUG_OBJECT (sink, "free connection");
gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
}
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
}
}
static GstRTSPResult
gst_rtsp_client_sink_open (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPResult ret;
sink->methods =
GST_RTSP_SETUP | GST_RTSP_RECORD | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
g_mutex_lock (&sink->open_conn_lock);
sink->open_conn_start = TRUE;
g_cond_broadcast (&sink->open_conn_cond);
GST_DEBUG_OBJECT (sink, "connection to server started");
g_mutex_unlock (&sink->open_conn_lock);
if ((ret = gst_rtsp_client_sink_connect_to_server (sink, async)) < 0)
goto open_failed;
if (async)
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret);
return ret;
/* ERRORS */
open_failed:
{
GST_WARNING_OBJECT (sink, "Failed to connect to server");
sink->open_error = TRUE;
if (async)
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret);
return ret;
}
}
static GstRTSPResult
gst_rtsp_client_sink_close (GstRTSPClientSink * sink, gboolean async,
gboolean only_close)
{
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPResult res = GST_RTSP_OK;
GList *walk;
const gchar *control;
GST_DEBUG_OBJECT (sink, "TEARDOWN...");
gst_rtsp_client_sink_set_state (sink, GST_STATE_NULL);
if (sink->state < GST_RTSP_STATE_READY) {
GST_DEBUG_OBJECT (sink, "not ready, doing cleanup");
goto close;
}
if (only_close)
goto close;
/* construct a control url */
control = get_aggregate_control (sink);
if (!(sink->methods & (GST_RTSP_RECORD | GST_RTSP_TEARDOWN)))
goto not_supported;
/* stop streaming */
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
if (context->stream_transport) {
gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
gst_object_unref (context->stream_transport);
context->stream_transport = NULL;
}
if (context->joined) {
gst_rtsp_stream_leave_bin (context->stream, GST_BIN (sink->internal_bin),
sink->rtpbin);
context->joined = FALSE;
}
}
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
const gchar *setup_url;
GstRTSPConnInfo *info;
GST_DEBUG_OBJECT (sink, "Looking at stream %p for teardown",
context->stream);
/* try aggregate control first but do non-aggregate control otherwise */
if (control)
setup_url = control;
else if ((setup_url = context->conninfo.location) == NULL) {
GST_DEBUG_OBJECT (sink, "Skipping TEARDOWN stream %p - no setup URL",
context->stream);
continue;
}
if (sink->conninfo.connection) {
info = &sink->conninfo;
} else if (context->conninfo.connection) {
info = &context->conninfo;
} else {
continue;
}
if (!info->connected)
goto next;
/* do TEARDOWN */
GST_DEBUG_OBJECT (sink, "Sending teardown for stream %p at URL %s",
context->stream, setup_url);
res =
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_TEARDOWN,
setup_url);
if (res < 0)
goto create_request_failed;
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "close", ("Closing stream"));
if ((res =
gst_rtsp_client_sink_send (sink, info, &request,
&response, NULL)) < 0)
goto send_error;
/* FIXME, parse result? */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
next:
/* early exit when we did aggregate control */
if (control)
break;
}
close:
/* close connections */
GST_DEBUG_OBJECT (sink, "closing connection...");
gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
gst_rtsp_conninfo_close (sink, &stream->conninfo, TRUE);
}
/* cleanup */
gst_rtsp_client_sink_cleanup (sink);
sink->state = GST_RTSP_STATE_INVALID;
if (async)
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_CLOSE, res);
return res;
/* ERRORS */
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto close;
}
send_error:
{
gchar *str = gst_rtsp_strresult (res);
gst_rtsp_message_unset (&request);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "TEARDOWN interrupted");
}
g_free (str);
goto close;
}
not_supported:
{
GST_DEBUG_OBJECT (sink,
"TEARDOWN and PLAY not supported, can't do TEARDOWN");
goto close;
}
}
static gboolean
gst_rtsp_client_sink_configure_manager (GstRTSPClientSink * sink)
{
GstElement *rtpbin;
GstStateChangeReturn ret;
rtpbin = sink->rtpbin;
if (rtpbin == NULL) {
GObjectClass *klass;
rtpbin = gst_element_factory_make ("rtpbin", NULL);
if (rtpbin == NULL)
goto no_rtpbin;
gst_bin_add (GST_BIN_CAST (sink->internal_bin), rtpbin);
sink->rtpbin = rtpbin;
/* Any more settings we should configure on rtpbin here? */
g_object_set (sink->rtpbin, "latency", sink->latency, NULL);
klass = G_OBJECT_GET_CLASS (G_OBJECT (rtpbin));
if (g_object_class_find_property (klass, "ntp-time-source")) {
g_object_set (sink->rtpbin, "ntp-time-source", sink->ntp_time_source,
NULL);
}
if (sink->sdes && g_object_class_find_property (klass, "sdes")) {
g_object_set (sink->rtpbin, "sdes", sink->sdes, NULL);
}
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_NEW_MANAGER], 0,
sink->rtpbin);
}
ret = gst_element_set_state (rtpbin, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_manager_failure;
return TRUE;
no_rtpbin:
{
GST_WARNING ("no rtpbin element");
g_warning ("failed to create element 'rtpbin', check your installation");
return FALSE;
}
start_manager_failure:
{
GST_DEBUG_OBJECT (sink, "could not start session manager");
gst_bin_remove (GST_BIN_CAST (sink->internal_bin), rtpbin);
return FALSE;
}
}
static GstElement *
request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPClientSink * sink)
{
GstRTSPStream *stream = NULL;
GstElement *ret = NULL;
GList *walk;
GST_RTSP_STATE_LOCK (sink);
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
if (sessid == gst_rtsp_stream_get_index (context->stream)) {
stream = context->stream;
break;
}
}
if (stream != NULL) {
GST_DEBUG_OBJECT (sink, "Creating aux sender for stream %u", sessid);
ret = gst_rtsp_stream_request_aux_sender (stream, sessid);
}
GST_RTSP_STATE_UNLOCK (sink);
return ret;
}
static GstElement *
request_fec_encoder (GstElement * rtpbin, guint sessid,
GstRTSPClientSink * sink)
{
GstRTSPStream *stream = NULL;
GstElement *ret = NULL;
GList *walk;
GST_RTSP_STATE_LOCK (sink);
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
if (sessid == gst_rtsp_stream_get_index (context->stream)) {
stream = context->stream;
break;
}
}
if (stream != NULL) {
ret = gst_rtsp_stream_request_ulpfec_encoder (stream, sessid);
}
GST_RTSP_STATE_UNLOCK (sink);
return ret;
}
static gboolean
gst_rtsp_client_sink_is_stopping (GstRTSPClientSink * sink)
{
gboolean is_stopping;
GST_OBJECT_LOCK (sink);
is_stopping = sink->task == NULL;
GST_OBJECT_UNLOCK (sink);
return is_stopping;
}
static gboolean
gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink)
{
GstRTSPStreamContext *context;
GList *walk;
const gchar *base;
gchar *stream_path;
GstUri *base_uri, *uri;
GST_DEBUG_OBJECT (sink, "Collecting stream information");
if (!gst_rtsp_client_sink_configure_manager (sink))
return FALSE;
base = get_aggregate_control (sink);
base_uri = gst_uri_from_string (base);
if (!base_uri) {
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, (NULL),
("Could not parse uri %s", base));
return FALSE;
}
g_mutex_lock (&sink->preroll_lock);
while (sink->contexts == NULL && !sink->conninfo.flushing) {
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
}
g_mutex_unlock (&sink->preroll_lock);
/* FIXME: Need different locking - need to protect against pad releases
* and potential state changes ruining things here */
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstPad *srcpad;
context = (GstRTSPStreamContext *) walk->data;
if (context->stream)
continue;
g_mutex_lock (&sink->preroll_lock);
while (!context->prerolled && !sink->conninfo.flushing
&& !gst_rtsp_client_sink_is_stopping (sink)) {
GST_DEBUG_OBJECT (sink, "Waiting for caps on stream %d", context->index);
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
}
if (sink->conninfo.flushing) {
g_mutex_unlock (&sink->preroll_lock);
break;
}
g_mutex_unlock (&sink->preroll_lock);
if (context->payloader == NULL)
continue;
srcpad = gst_element_get_static_pad (context->payloader, "src");
GST_DEBUG_OBJECT (sink, "Creating stream object for stream %d",
context->index);
context->stream =
gst_rtsp_client_sink_create_stream (sink, context, context->payloader,
srcpad);
/* append stream index to uri path */
g_free (context->conninfo.location);
stream_path = g_strdup_printf ("stream=%d", context->index);
uri = gst_uri_copy (base_uri);
gst_uri_append_path (uri, stream_path);
context->conninfo.location = gst_uri_to_string (uri);
gst_uri_unref (uri);
g_free (stream_path);
if (sink->rtx_time > 0) {
/* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
g_signal_connect (sink->rtpbin, "request-aux-sender",
(GCallback) request_aux_sender, sink);
}
g_signal_connect (sink->rtpbin, "request-fec-encoder",
(GCallback) request_fec_encoder, sink);
if (!gst_rtsp_stream_join_bin (context->stream,
GST_BIN (sink->internal_bin), sink->rtpbin, GST_STATE_PAUSED)) {
goto join_bin_failed;
}
context->joined = TRUE;
/* Block the stream, as it does not have any transport parts yet */
gst_rtsp_stream_set_blocked (context->stream, TRUE);
/* Let the stream object receive data */
gst_pad_remove_probe (srcpad, context->payloader_block_id);
gst_object_unref (srcpad);
}
/* Now wait for the preroll of the rtp bin */
g_mutex_lock (&sink->preroll_lock);
while (!sink->prerolled && sink->conninfo.connection
&& !sink->conninfo.flushing) {
GST_LOG_OBJECT (sink, "Waiting for preroll before continuing");
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
}
GST_LOG_OBJECT (sink, "Marking streams as collected");
sink->streams_collected = TRUE;
g_mutex_unlock (&sink->preroll_lock);
gst_uri_unref (base_uri);
return TRUE;
join_bin_failed:
gst_uri_unref (base_uri);
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not start stream %d", context->index));
return FALSE;
}
static GstRTSPResult
gst_rtsp_client_sink_create_transports_string (GstRTSPClientSink * sink,
GstRTSPStreamContext * context, GSocketFamily family,
GstRTSPLowerTrans protocols, GstRTSPProfile profiles, gchar ** transports)
{
GString *result;
GstRTSPStream *stream = context->stream;
gboolean first = TRUE;
/* the default RTSP transports */
result = g_string_new ("RTP");
while (profiles != 0) {
if (!first)
g_string_append (result, ",RTP");
if (profiles & GST_RTSP_PROFILE_SAVPF) {
g_string_append (result, "/SAVPF");
profiles &= ~GST_RTSP_PROFILE_SAVPF;
} else if (profiles & GST_RTSP_PROFILE_SAVP) {
g_string_append (result, "/SAVP");
profiles &= ~GST_RTSP_PROFILE_SAVP;
} else if (profiles & GST_RTSP_PROFILE_AVPF) {
g_string_append (result, "/AVPF");
profiles &= ~GST_RTSP_PROFILE_AVPF;
} else if (profiles & GST_RTSP_PROFILE_AVP) {
g_string_append (result, "/AVP");
profiles &= ~GST_RTSP_PROFILE_AVP;
} else {
GST_WARNING_OBJECT (sink, "Unimplemented profile(s) 0x%x", profiles);
break;
}
if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
GstRTSPRange ports;
GST_DEBUG_OBJECT (sink, "adding UDP unicast");
gst_rtsp_stream_get_server_port (stream, &ports, family);
g_string_append_printf (result, "/UDP;unicast;client_port=%d-%d",
ports.min, ports.max);
} else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
GstRTSPAddress *addr =
gst_rtsp_stream_get_multicast_address (stream, family);
if (addr) {
GST_DEBUG_OBJECT (sink, "adding UDP multicast");
g_string_append_printf (result, "/UDP;multicast;client_port=%d-%d",
addr->port, addr->port + addr->n_ports - 1);
gst_rtsp_address_free (addr);
}
} else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
GST_DEBUG_OBJECT (sink, "adding TCP");
g_string_append_printf (result, "/TCP;unicast;interleaved=%d-%d",
sink->free_channel, sink->free_channel + 1);
}
g_string_append (result, ";mode=RECORD");
/* FIXME: Support appending too:
if (sink->append)
g_string_append (result, ";append");
*/
first = FALSE;
}
if (first) {
/* No valid transport could be constructed */
GST_ERROR_OBJECT (sink, "No supported profiles configured");
goto fail;
}
*transports = g_string_free (result, FALSE);
GST_DEBUG_OBJECT (sink, "prepared transports %s", GST_STR_NULL (*transports));
return GST_RTSP_OK;
fail:
g_string_free (result, TRUE);
return GST_RTSP_ERROR;
}
static GstCaps *
signal_get_srtcp_params (GstRTSPClientSink * sink,
GstRTSPStreamContext * context)
{
GstCaps *caps = NULL;
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_REQUEST_RTCP_KEY], 0,
context->index, &caps);
if (caps != NULL)
GST_DEBUG_OBJECT (sink, "SRTP parameters received");
return caps;
}
static gchar *
gst_rtsp_client_sink_stream_make_keymgmt (GstRTSPClientSink * sink,
GstRTSPStreamContext * context)
{
gchar *base64, *result = NULL;
GstMIKEYMessage *mikey_msg;
context->srtcpparams = signal_get_srtcp_params (sink, context);
if (context->srtcpparams == NULL)
context->srtcpparams = gst_rtsp_stream_get_caps (context->stream);
mikey_msg = gst_mikey_message_new_from_caps (context->srtcpparams);
if (mikey_msg) {
guint send_ssrc, send_rtx_ssrc;
const GstStructure *s = gst_caps_get_structure (context->srtcpparams, 0);
/* add policy '0' for our SSRC */
gst_rtsp_stream_get_ssrc (context->stream, &send_ssrc);
GST_LOG_OBJECT (sink, "Stream %p ssrc %x", context->stream, send_ssrc);
gst_mikey_message_add_cs_srtp (mikey_msg, 0, send_ssrc, 0);
if (gst_structure_get_uint (s, "rtx-ssrc", &send_rtx_ssrc))
gst_mikey_message_add_cs_srtp (mikey_msg, 0, send_rtx_ssrc, 0);
base64 = gst_mikey_message_base64_encode (mikey_msg);
gst_mikey_message_unref (mikey_msg);
if (base64) {
result = gst_sdp_make_keymgmt (context->conninfo.location, base64);
g_free (base64);
}
}
return result;
}
/* masks to be kept in sync with the hardcoded protocol order of preference
* in code below */
static const guint protocol_masks[] = {
GST_RTSP_LOWER_TRANS_UDP,
GST_RTSP_LOWER_TRANS_UDP_MCAST,
GST_RTSP_LOWER_TRANS_TCP,
0
};
/* Same for profile_masks */
static const guint profile_masks[] = {
GST_RTSP_PROFILE_SAVPF,
GST_RTSP_PROFILE_SAVP,
GST_RTSP_PROFILE_AVPF,
GST_RTSP_PROFILE_AVP,
0
};
static gboolean
do_send_data (GstBuffer * buffer, guint8 channel,
GstRTSPStreamContext * context)
{
GstRTSPClientSink *sink = context->parent;
GstRTSPMessage message = { 0 };
GstRTSPResult res = GST_RTSP_OK;
gst_rtsp_message_init_data (&message, channel);
gst_rtsp_message_set_body_buffer (&message, buffer);
res =
gst_rtsp_client_sink_try_send (sink, &sink->conninfo, &message, 1,
NULL, NULL);
gst_rtsp_message_unset (&message);
gst_rtsp_stream_transport_message_sent (context->stream_transport);
return res == GST_RTSP_OK;
}
static gboolean
do_send_data_list (GstBufferList * buffer_list, guint8 channel,
GstRTSPStreamContext * context)
{
GstRTSPClientSink *sink = context->parent;
GstRTSPResult res = GST_RTSP_OK;
guint i, n = gst_buffer_list_length (buffer_list);
GstRTSPMessage *messages = g_newa (GstRTSPMessage, n);
memset (messages, 0, n * sizeof (GstRTSPMessage));
for (i = 0; i < n; i++) {
GstBuffer *buffer = gst_buffer_list_get (buffer_list, i);
gst_rtsp_message_init_data (&messages[i], channel);
gst_rtsp_message_set_body_buffer (&messages[i], buffer);
}
res =
gst_rtsp_client_sink_try_send (sink, &sink->conninfo, messages, n,
NULL, NULL);
for (i = 0; i < n; i++) {
gst_rtsp_message_unset (&messages[i]);
gst_rtsp_stream_transport_message_sent (context->stream_transport);
}
return res == GST_RTSP_OK;
}
static GstRTSPResult
gst_rtsp_client_sink_setup_streams (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPResult res = GST_RTSP_ERROR;
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPLowerTrans protocols;
GstRTSPStatusCode code;
GSocketFamily family;
GSocketAddress *sa;
GSocket *conn_socket;
GstRTSPUrl *url;
GList *walk;
gchar *hval;
if (sink->conninfo.connection) {
url = gst_rtsp_connection_get_url (sink->conninfo.connection);
/* we initially allow all configured lower transports. based on the URL
* transports and the replies from the server we narrow them down. */
protocols = url->transports & sink->cur_protocols;
} else {
url = NULL;
protocols = sink->cur_protocols;
}
if (protocols == 0)
goto no_protocols;
GST_RTSP_STATE_LOCK (sink);
if (G_UNLIKELY (sink->contexts == NULL))
goto no_streams;
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
GstRTSPStream *stream;
GstRTSPConnInfo *info;
GstRTSPProfile profiles;
GstRTSPProfile cur_profile;
gchar *transports;
gint retry = 0;
guint profile_mask = 0;
guint mask = 0;
GstCaps *caps;
const GstSDPMedia *media;
stream = context->stream;
profiles = gst_rtsp_stream_get_profiles (stream);
caps = gst_rtsp_stream_get_caps (stream);
if (caps == NULL) {
GST_DEBUG_OBJECT (sink, "skipping stream %p, no caps", stream);
continue;
}
gst_caps_unref (caps);
media = gst_sdp_message_get_media (&sink->cursdp, context->sdp_index);
if (media == NULL) {
GST_DEBUG_OBJECT (sink, "skipping stream %p, no SDP info", stream);
continue;
}
/* skip setup if we have no URL for it */
if (context->conninfo.location == NULL) {
GST_DEBUG_OBJECT (sink, "skipping stream %p, no setup", stream);
continue;
}
if (sink->conninfo.connection == NULL) {
if (!gst_rtsp_conninfo_connect (sink, &context->conninfo, async)) {
GST_DEBUG_OBJECT (sink, "skipping stream %p, failed to connect",
stream);
continue;
}
info = &context->conninfo;
} else {
info = &sink->conninfo;
}
GST_DEBUG_OBJECT (sink, "doing setup of stream %p with %s", stream,
context->conninfo.location);
conn_socket = gst_rtsp_connection_get_read_socket (info->connection);
sa = g_socket_get_local_address (conn_socket, NULL);
family = g_socket_address_get_family (sa);
g_object_unref (sa);
next_protocol:
/* first selectable profile */
while (profile_masks[profile_mask]
&& !(profiles & profile_masks[profile_mask]))
profile_mask++;
if (!profile_masks[profile_mask])
goto no_profiles;
/* first selectable protocol */
while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
mask++;
if (!protocol_masks[mask])
goto no_protocols;
retry:
GST_DEBUG_OBJECT (sink, "protocols = 0x%x, protocol mask = 0x%x", protocols,
protocol_masks[mask]);
/* create a string with first transport in line */
transports = NULL;
cur_profile = profiles & profile_masks[profile_mask];
res = gst_rtsp_client_sink_create_transports_string (sink, context, family,
protocols & protocol_masks[mask], cur_profile, &transports);
if (res < 0 || transports == NULL)
goto setup_transport_failed;
if (strlen (transports) == 0) {
g_free (transports);
GST_DEBUG_OBJECT (sink, "no transports found");
mask++;
profile_mask = 0;
goto next_protocol;
}
GST_DEBUG_OBJECT (sink, "transport is %s", GST_STR_NULL (transports));
/* create SETUP request */
res =
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_SETUP,
context->conninfo.location);
if (res < 0) {
g_free (transports);
goto create_request_failed;
}
/* set up keys */
if (cur_profile == GST_RTSP_PROFILE_SAVP ||
cur_profile == GST_RTSP_PROFILE_SAVPF) {
hval = gst_rtsp_client_sink_stream_make_keymgmt (sink, context);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval);
}
/* if the user wants a non default RTP packet size we add the blocksize
* parameter */
if (sink->rtp_blocksize > 0) {
hval = g_strdup_printf ("%d", sink->rtp_blocksize);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
}
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "request", ("SETUP stream %d",
context->index));
{
GstRTSPTransport *transport;
gst_rtsp_transport_new (&transport);
if (gst_rtsp_transport_parse (transports, transport) != GST_RTSP_OK)
goto parse_transport_failed;
if (transport->lower_transport != GST_RTSP_LOWER_TRANS_TCP) {
if (!gst_rtsp_stream_allocate_udp_sockets (stream, family, transport,
FALSE)) {
gst_rtsp_transport_free (transport);
goto allocate_udp_ports_failed;
}
}
if (!gst_rtsp_stream_complete_stream (stream, transport)) {
gst_rtsp_transport_free (transport);
goto complete_stream_failed;
}
gst_rtsp_transport_free (transport);
gst_rtsp_stream_set_blocked (stream, FALSE);
}
/* FIXME:
* the creation of the transports string depends on
* calling stream_get_server_port, which only starts returning
* something meaningful after a call to stream_allocate_udp_sockets
* has been made, this function expects a transport that we parse
* from the transport string ...
*
* Significant refactoring is in order, but does not look entirely
* trivial, for now we put a band aid on and create a second transport
* string after the stream has been completed, to pass it in
* the request headers instead of the previous, incomplete one.
*/
g_free (transports);
transports = NULL;
res = gst_rtsp_client_sink_create_transports_string (sink, context, family,
protocols & protocol_masks[mask], cur_profile, &transports);
if (res < 0 || transports == NULL)
goto setup_transport_failed;
/* select transport */
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
/* handle the code ourselves */
res = gst_rtsp_client_sink_send (sink, info, &request, &response, &code);
if (res < 0)
goto send_error;
switch (code) {
case GST_RTSP_STS_OK:
break;
case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
/* Try another profile. If no more, move to the next protocol */
profile_mask++;
while (profile_masks[profile_mask]
&& !(profiles & profile_masks[profile_mask]))
profile_mask++;
if (profile_masks[profile_mask])
goto retry;
/* select next available protocol, give up on this stream if none */
/* Reset profiles to try: */
profile_mask = 0;
mask++;
while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
mask++;
if (!protocol_masks[mask])
continue;
else
goto retry;
default:
goto response_error;
}
/* parse response transport */
{
gchar *resptrans = NULL;
GstRTSPTransport *transport;
gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
&resptrans, 0);
if (!resptrans) {
goto no_transport;
}
gst_rtsp_transport_new (&transport);
/* parse transport, go to next stream on parse error */
if (gst_rtsp_transport_parse (resptrans, transport) != GST_RTSP_OK) {
GST_WARNING_OBJECT (sink, "failed to parse transport %s", resptrans);
goto next;
}
/* update allowed transports for other streams. once the transport of
* one stream has been determined, we make sure that all other streams
* are configured in the same way */
switch (transport->lower_transport) {
case GST_RTSP_LOWER_TRANS_TCP:
GST_DEBUG_OBJECT (sink, "stream %p as TCP interleaved", stream);
protocols = GST_RTSP_LOWER_TRANS_TCP;
sink->interleaved = TRUE;
/* update free channels */
sink->free_channel =
MAX (transport->interleaved.min, sink->free_channel);
sink->free_channel =
MAX (transport->interleaved.max, sink->free_channel);
sink->free_channel++;
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
/* only allow multicast for other streams */
GST_DEBUG_OBJECT (sink, "stream %p as UDP multicast", stream);
protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
break;
case GST_RTSP_LOWER_TRANS_UDP:
/* only allow unicast for other streams */
GST_DEBUG_OBJECT (sink, "stream %p as UDP unicast", stream);
protocols = GST_RTSP_LOWER_TRANS_UDP;
/* Update transport with server destination if not provided by the server */
if (transport->destination == NULL) {
transport->destination = g_strdup (sink->server_ip);
}
break;
default:
GST_DEBUG_OBJECT (sink, "stream %p unknown transport %d", stream,
transport->lower_transport);
break;
}
if (!retry) {
GST_DEBUG ("Configuring the stream transport for stream %d",
context->index);
if (context->stream_transport == NULL)
context->stream_transport =
gst_rtsp_stream_transport_new (stream, transport);
else
gst_rtsp_stream_transport_set_transport (context->stream_transport,
transport);
if (transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* our callbacks to send data on this TCP connection */
gst_rtsp_stream_transport_set_callbacks (context->stream_transport,
(GstRTSPSendFunc) do_send_data,
(GstRTSPSendFunc) do_send_data, context, NULL);
gst_rtsp_stream_transport_set_list_callbacks
(context->stream_transport,
(GstRTSPSendListFunc) do_send_data_list,
(GstRTSPSendListFunc) do_send_data_list, context, NULL);
}
/* The stream_transport now owns the transport */
transport = NULL;
gst_rtsp_stream_transport_set_active (context->stream_transport, TRUE);
}
next:
if (transport)
gst_rtsp_transport_free (transport);
/* clean up used RTSP messages */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
}
}
GST_RTSP_STATE_UNLOCK (sink);
/* store the transport protocol that was configured */
sink->cur_protocols = protocols;
return res;
no_streams:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("SDP contains no streams"));
return GST_RTSP_ERROR;
}
setup_transport_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not setup transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
no_profiles:
{
GST_RTSP_STATE_UNLOCK (sink);
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not connect to server, no profiles left"));
return GST_RTSP_ERROR;
}
no_protocols:
{
GST_RTSP_STATE_UNLOCK (sink);
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not connect to server, no protocols left"));
return GST_RTSP_ERROR;
}
no_transport:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Server did not select transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto cleanup_error;
}
parse_transport_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not parse transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
allocate_udp_ports_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not parse transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
complete_stream_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not parse transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
send_error:
{
gchar *str = gst_rtsp_strresult (res);
GST_RTSP_STATE_UNLOCK (sink);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "send interrupted");
}
g_free (str);
goto cleanup_error;
}
response_error:
{
const gchar *str = gst_rtsp_status_as_text (code);
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Error (%d): %s", code, GST_STR_NULL (str)));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
cleanup_error:
{
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
}
}
static GstRTSPResult
gst_rtsp_client_sink_ensure_open (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPResult res = GST_RTSP_OK;
if (sink->state < GST_RTSP_STATE_READY) {
res = GST_RTSP_ERROR;
if (sink->open_error) {
GST_DEBUG_OBJECT (sink, "the stream was in error");
goto done;
}
if (async)
gst_rtsp_client_sink_loop_start_cmd (sink, CMD_OPEN);
if ((res = gst_rtsp_client_sink_open (sink, async)) < 0) {
GST_DEBUG_OBJECT (sink, "failed to open stream");
goto done;
}
}
done:
return res;
}
static GstRTSPResult
gst_rtsp_client_sink_record (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPResult res = GST_RTSP_OK;
GstSDPMessage *sdp;
guint sdp_index = 0;
GstSDPInfo info = { 0, };
gchar *keymgmt;
guint i;
const gchar *proto;
gchar *sess_id, *client_ip, *str;
GSocketAddress *sa;
GInetAddress *ia;
GSocket *conn_socket;
GList *walk;
g_mutex_lock (&sink->preroll_lock);
if (sink->state == GST_RTSP_STATE_PLAYING) {
/* Already recording, don't send another request */
GST_LOG_OBJECT (sink, "Already in RECORD. Skipping duplicate request.");
g_mutex_unlock (&sink->preroll_lock);
goto done;
}
g_mutex_unlock (&sink->preroll_lock);
/* Collect all our input streams and create
* stream objects before actually returning.
* The streams are blocked at this point as we do not have any transport
* parts yet. */
gst_rtsp_client_sink_collect_streams (sink);
if (gst_rtsp_client_sink_is_stopping (sink)) {
GST_INFO_OBJECT (sink, "task stopped, shutting down");
return GST_RTSP_EINTR;
}
g_mutex_lock (&sink->block_streams_lock);
/* Wait for streams to be blocked */
while (sink->n_streams_blocked < g_list_length (sink->contexts)
&& !gst_rtsp_client_sink_is_stopping (sink)) {
GST_DEBUG_OBJECT (sink, "waiting for streams to be blocked");
g_cond_wait (&sink->block_streams_cond, &sink->block_streams_lock);
}
g_mutex_unlock (&sink->block_streams_lock);
if (gst_rtsp_client_sink_is_stopping (sink)) {
GST_INFO_OBJECT (sink, "task stopped, shutting down");
return GST_RTSP_EINTR;
}
/* Send announce, then setup for all streams */
gst_sdp_message_init (&sink->cursdp);
sdp = &sink->cursdp;
/* some standard things first */
gst_sdp_message_set_version (sdp, "0");
/* session ID doesn't have to be super-unique in this case */
sess_id = g_strdup_printf ("%u", g_random_int ());
if (sink->conninfo.connection == NULL)
return GST_RTSP_ERROR;
conn_socket = gst_rtsp_connection_get_read_socket (sink->conninfo.connection);
sa = g_socket_get_local_address (conn_socket, NULL);
ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
client_ip = g_inet_address_to_string (ia);
if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV6) {
info.is_ipv6 = TRUE;
proto = "IP6";
} else if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV4)
proto = "IP4";
else
g_assert_not_reached ();
g_object_unref (sa);
/* FIXME: Should this actually be the server's IP or ours? */
info.server_ip = sink->server_ip;
gst_sdp_message_set_origin (sdp, "-", sess_id, "1", "IN", proto, client_ip);
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
gst_sdp_message_set_information (sdp, "rtspclientsink");
gst_sdp_message_add_time (sdp, "0", "0", NULL);
gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
/* add stream */
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
gst_rtsp_sdp_from_stream (sdp, &info, context->stream);
context->sdp_index = sdp_index++;
}
g_free (sess_id);
g_free (client_ip);
/* send ANNOUNCE request */
GST_DEBUG_OBJECT (sink, "create ANNOUNCE request...");
res =
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_ANNOUNCE,
sink->conninfo.url_str);
if (res < 0)
goto create_request_failed;
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_UPDATE_SDP], 0, sdp);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CONTENT_TYPE,
"application/sdp");
/* add SDP to the request body */
str = gst_sdp_message_as_text (sdp);
gst_rtsp_message_take_body (&request, (guint8 *) str, strlen (str));
/* send ANNOUNCE */
GST_DEBUG_OBJECT (sink, "sending announce...");
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "record",
("Sending server stream info"));
if ((res =
gst_rtsp_client_sink_send (sink, &sink->conninfo, &request,
&response, NULL)) < 0)
goto send_error;
/* parse the keymgmt */
i = 0;
walk = sink->contexts;
while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_KEYMGMT,
&keymgmt, i++) == GST_RTSP_OK) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
walk = g_list_next (walk);
if (!gst_rtsp_stream_handle_keymgmt (context->stream, keymgmt))
goto keymgmt_error;
}
/* send setup for all streams */
if ((res = gst_rtsp_client_sink_setup_streams (sink, async)) < 0)
goto setup_failed;
res = gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_RECORD,
sink->conninfo.url_str);
if (res < 0)
goto create_request_failed;
#if 0 /* FIXME: Configure a range based on input segments? */
if (src->need_range) {
hval = gen_range_header (src, segment);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
}
if (segment->rate != 1.0) {
gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
g_ascii_dtostr (hval, sizeof (hval), segment->rate);
if (src->skip)
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
else
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
}
#endif
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "record", ("Starting recording"));
if ((res =
gst_rtsp_client_sink_send (sink, &sink->conninfo, &request,
&response, NULL)) < 0)
goto send_error;
#if 0 /* FIXME: Check if servers return these for record: */
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
* for the RTP packets. If this is not present, we assume all starts from 0...
* This is info for the RTP session manager that we pass to it in caps. */
hval_idx = 0;
while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
&hval, hval_idx++) == GST_RTSP_OK)
gst_rtspsrc_parse_rtpinfo (src, hval);
/* some servers indicate RTCP parameters in PLAY response,
* rather than properly in SDP */
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
&hval, 0) == GST_RTSP_OK)
gst_rtspsrc_handle_rtcp_interval (src, hval);
#endif
gst_rtsp_client_sink_set_state (sink, GST_STATE_PLAYING);
sink->state = GST_RTSP_STATE_PLAYING;
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
gst_rtsp_stream_unblock_rtcp (context->stream);
}
/* clean up any messages */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
done:
return res;
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto cleanup_error;
}
send_error:
{
/* Don't post a message - the rtsp_send method will have
* taken care of it because we passed NULL for the response code */
goto cleanup_error;
}
keymgmt_error:
{
GST_ELEMENT_ERROR (sink, STREAM, DECRYPT_NOKEY, (NULL),
("Could not handle KeyMgmt"));
}
setup_failed:
{
GST_ERROR_OBJECT (sink, "setup failed");
goto cleanup_error;
}
cleanup_error:
{
if (sink->conninfo.connection) {
GST_DEBUG_OBJECT (sink, "free connection");
gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
}
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
}
}
static GstRTSPResult
gst_rtsp_client_sink_pause (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPResult res = GST_RTSP_OK;
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GList *walk;
const gchar *control;
GST_DEBUG_OBJECT (sink, "PAUSE...");
if ((res = gst_rtsp_client_sink_ensure_open (sink, async)) < 0)
goto open_failed;
if (!(sink->methods & GST_RTSP_PAUSE))
goto not_supported;
if (sink->state == GST_RTSP_STATE_READY)
goto was_paused;
if (!sink->conninfo.connection || !sink->conninfo.connected)
goto no_connection;
/* construct a control url */
control = get_aggregate_control (sink);
/* loop over the streams. We might exit the loop early when we could do an
* aggregate control */
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
GstRTSPConnInfo *info;
const gchar *setup_url;
/* try aggregate control first but do non-aggregate control otherwise */
if (control)
setup_url = control;
else if ((setup_url = stream->conninfo.location) == NULL)
continue;
if (sink->conninfo.connection) {
info = &sink->conninfo;
} else if (stream->conninfo.connection) {
info = &stream->conninfo;
} else {
continue;
}
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "request",
("Sending PAUSE request"));
if ((res =
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_PAUSE,
setup_url)) < 0)
goto create_request_failed;
if ((res =
gst_rtsp_client_sink_send (sink, info, &request, &response,
NULL)) < 0)
goto send_error;
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
/* exit early when we did agregate control */
if (control)
break;
}
/* change element states now */
gst_rtsp_client_sink_set_state (sink, GST_STATE_PAUSED);
no_connection:
sink->state = GST_RTSP_STATE_READY;
done:
if (async)
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_PAUSE, res);
return res;
/* ERRORS */
open_failed:
{
GST_DEBUG_OBJECT (sink, "failed to open stream");
goto done;
}
not_supported:
{
GST_DEBUG_OBJECT (sink, "PAUSE is not supported");
goto done;
}
was_paused:
{
GST_DEBUG_OBJECT (sink, "we were already PAUSED");
goto done;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto done;
}
send_error:
{
gchar *str = gst_rtsp_strresult (res);
gst_rtsp_message_unset (&request);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "PAUSE interrupted");
}
g_free (str);
goto done;
}
}
static void
gst_rtsp_client_sink_handle_message (GstBin * bin, GstMessage * message)
{
GstRTSPClientSink *rtsp_client_sink;
rtsp_client_sink = GST_RTSP_CLIENT_SINK (bin);
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ELEMENT:
{
const GstStructure *s = gst_message_get_structure (message);
if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
gboolean ignore_timeout;
GST_DEBUG_OBJECT (bin, "timeout on UDP port");
GST_OBJECT_LOCK (rtsp_client_sink);
ignore_timeout = rtsp_client_sink->ignore_timeout;
rtsp_client_sink->ignore_timeout = TRUE;
GST_OBJECT_UNLOCK (rtsp_client_sink);
/* we only act on the first udp timeout message, others are irrelevant
* and can be ignored. */
if (!ignore_timeout)
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_RECONNECT,
CMD_LOOP);
/* eat and free */
gst_message_unref (message);
return;
} else if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
/* An RTSPStream has prerolled */
GST_DEBUG_OBJECT (rtsp_client_sink, "received GstRTSPStreamBlocking");
g_mutex_lock (&rtsp_client_sink->block_streams_lock);
rtsp_client_sink->n_streams_blocked++;
g_cond_broadcast (&rtsp_client_sink->block_streams_cond);
g_mutex_unlock (&rtsp_client_sink->block_streams_lock);
}
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
case GST_MESSAGE_ASYNC_START:{
GstObject *sender;
sender = GST_MESSAGE_SRC (message);
GST_LOG_OBJECT (rtsp_client_sink,
"Have async-start from %" GST_PTR_FORMAT, sender);
if (sender == GST_OBJECT (rtsp_client_sink->internal_bin)) {
GST_LOG_OBJECT (rtsp_client_sink, "child bin is now ASYNC");
}
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
case GST_MESSAGE_ASYNC_DONE:
{
GstObject *sender;
gboolean need_async_done;
sender = GST_MESSAGE_SRC (message);
GST_LOG_OBJECT (rtsp_client_sink, "Have async-done from %" GST_PTR_FORMAT,
sender);
g_mutex_lock (&rtsp_client_sink->preroll_lock);
if (sender == GST_OBJECT_CAST (rtsp_client_sink->internal_bin)) {
GST_LOG_OBJECT (rtsp_client_sink, "child bin is no longer ASYNC");
}
need_async_done = rtsp_client_sink->in_async;
if (rtsp_client_sink->in_async) {
rtsp_client_sink->in_async = FALSE;
g_cond_broadcast (&rtsp_client_sink->preroll_cond);
}
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
if (need_async_done) {
GST_DEBUG_OBJECT (rtsp_client_sink, "Posting ASYNC-DONE");
gst_element_post_message (GST_ELEMENT_CAST (rtsp_client_sink),
gst_message_new_async_done (GST_OBJECT_CAST (rtsp_client_sink),
GST_CLOCK_TIME_NONE));
}
break;
}
case GST_MESSAGE_ERROR:
{
GstObject *sender;
sender = GST_MESSAGE_SRC (message);
GST_DEBUG_OBJECT (rtsp_client_sink, "got error from %s",
GST_ELEMENT_NAME (sender));
/* FIXME: Ignore errors on RTCP? */
/* fatal but not our message, forward */
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
case GST_MESSAGE_STATE_CHANGED:
{
if (GST_MESSAGE_SRC (message) ==
(GstObject *) rtsp_client_sink->internal_bin) {
GstState newstate, pending;
gst_message_parse_state_changed (message, NULL, &newstate, &pending);
g_mutex_lock (&rtsp_client_sink->preroll_lock);
rtsp_client_sink->prerolled = (newstate >= GST_STATE_PAUSED)
&& pending == GST_STATE_VOID_PENDING;
g_cond_broadcast (&rtsp_client_sink->preroll_cond);
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
GST_DEBUG_OBJECT (bin,
"Internal bin changed state to %s (pending %s). Prerolled now %d",
gst_element_state_get_name (newstate),
gst_element_state_get_name (pending), rtsp_client_sink->prerolled);
}
/* fallthrough */
}
default:
{
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
}
}
/* the thread where everything happens */
static void
gst_rtsp_client_sink_thread (GstRTSPClientSink * sink)
{
gint cmd;
GST_OBJECT_LOCK (sink);
cmd = sink->pending_cmd;
if (cmd == CMD_RECONNECT || cmd == CMD_RECORD || cmd == CMD_PAUSE
|| cmd == CMD_LOOP || cmd == CMD_OPEN)
sink->pending_cmd = CMD_LOOP;
else
sink->pending_cmd = CMD_WAIT;
GST_DEBUG_OBJECT (sink, "got command %s", cmd_to_string (cmd));
/* we got the message command, so ensure communication is possible again */
gst_rtsp_client_sink_connection_flush (sink, FALSE);
sink->busy_cmd = cmd;
GST_OBJECT_UNLOCK (sink);
switch (cmd) {
case CMD_OPEN:
if (gst_rtsp_client_sink_open (sink, TRUE) == GST_RTSP_ERROR)
gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT,
CMD_ALL & ~CMD_CLOSE);
break;
case CMD_RECORD:
gst_rtsp_client_sink_record (sink, TRUE);
break;
case CMD_PAUSE:
gst_rtsp_client_sink_pause (sink, TRUE);
break;
case CMD_CLOSE:
gst_rtsp_client_sink_close (sink, TRUE, FALSE);
break;
case CMD_LOOP:
gst_rtsp_client_sink_loop (sink);
break;
case CMD_RECONNECT:
gst_rtsp_client_sink_reconnect (sink, FALSE);
break;
default:
break;
}
GST_OBJECT_LOCK (sink);
/* and go back to sleep */
if (sink->pending_cmd == CMD_WAIT) {
if (sink->task)
gst_task_pause (sink->task);
}
/* reset waiting */
sink->busy_cmd = CMD_WAIT;
GST_OBJECT_UNLOCK (sink);
}
static gboolean
gst_rtsp_client_sink_start (GstRTSPClientSink * sink)
{
GST_DEBUG_OBJECT (sink, "starting");
sink->streams_collected = FALSE;
gst_element_set_locked_state (GST_ELEMENT (sink->internal_bin), TRUE);
gst_rtsp_client_sink_set_state (sink, GST_STATE_READY);
GST_OBJECT_LOCK (sink);
sink->pending_cmd = CMD_WAIT;
if (sink->task == NULL) {
sink->task =
gst_task_new ((GstTaskFunction) gst_rtsp_client_sink_thread, sink,
NULL);
if (sink->task == NULL)
goto task_error;
gst_task_set_lock (sink->task, GST_RTSP_STREAM_GET_LOCK (sink));
}
GST_OBJECT_UNLOCK (sink);
return TRUE;
/* ERRORS */
task_error:
{
GST_OBJECT_UNLOCK (sink);
GST_ERROR_OBJECT (sink, "failed to create task");
return FALSE;
}
}
static gboolean
gst_rtsp_client_sink_stop (GstRTSPClientSink * sink)
{
GstTask *task;
GST_DEBUG_OBJECT (sink, "stopping");
/* also cancels pending task */
gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_ALL & ~CMD_CLOSE);
GST_OBJECT_LOCK (sink);
if ((task = sink->task)) {
sink->task = NULL;
GST_OBJECT_UNLOCK (sink);
gst_task_stop (task);
g_mutex_lock (&sink->block_streams_lock);
g_cond_broadcast (&sink->block_streams_cond);
g_mutex_unlock (&sink->block_streams_lock);
g_mutex_lock (&sink->preroll_lock);
g_cond_broadcast (&sink->preroll_cond);
g_mutex_unlock (&sink->preroll_lock);
/* make sure it is not running */
GST_RTSP_STREAM_LOCK (sink);
GST_RTSP_STREAM_UNLOCK (sink);
/* now wait for the task to finish */
gst_task_join (task);
/* and free the task */
gst_object_unref (GST_OBJECT (task));
GST_OBJECT_LOCK (sink);
}
GST_OBJECT_UNLOCK (sink);
/* ensure synchronously all is closed and clean */
gst_rtsp_client_sink_close (sink, FALSE, TRUE);
return TRUE;
}
static GstStateChangeReturn
gst_rtsp_client_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstRTSPClientSink *rtsp_client_sink;
GstStateChangeReturn ret;
rtsp_client_sink = GST_RTSP_CLIENT_SINK (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!gst_rtsp_client_sink_start (rtsp_client_sink))
goto start_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* init some state */
rtsp_client_sink->cur_protocols = rtsp_client_sink->protocols;
/* first attempt, don't ignore timeouts */
rtsp_client_sink->ignore_timeout = FALSE;
rtsp_client_sink->open_error = FALSE;
gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_PAUSED);
g_mutex_lock (&rtsp_client_sink->preroll_lock);
if (rtsp_client_sink->in_async) {
GST_DEBUG_OBJECT (rtsp_client_sink, "Posting ASYNC-START");
gst_element_post_message (GST_ELEMENT_CAST (rtsp_client_sink),
gst_message_new_async_start (GST_OBJECT_CAST (rtsp_client_sink)));
}
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
/* fall-through */
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* unblock the tcp tasks and make the loop waiting */
if (gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_WAIT,
CMD_LOOP)) {
/* make sure it is waiting before we send PLAY below */
GST_RTSP_STREAM_LOCK (rtsp_client_sink);
GST_RTSP_STREAM_UNLOCK (rtsp_client_sink);
}
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_READY);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
goto done;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* Return ASYNC and preroll input streams */
g_mutex_lock (&rtsp_client_sink->preroll_lock);
if (rtsp_client_sink->in_async)
ret = GST_STATE_CHANGE_ASYNC;
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_OPEN, 0);
/* CMD_OPEN has been scheduled. Wait until the sink thread starts
* opening connection to the server */
g_mutex_lock (&rtsp_client_sink->open_conn_lock);
while (!rtsp_client_sink->open_conn_start) {
GST_DEBUG_OBJECT (rtsp_client_sink,
"wait for connection to be started");
g_cond_wait (&rtsp_client_sink->open_conn_cond,
&rtsp_client_sink->open_conn_lock);
}
rtsp_client_sink->open_conn_start = FALSE;
g_mutex_unlock (&rtsp_client_sink->open_conn_lock);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
GST_DEBUG_OBJECT (rtsp_client_sink,
"Switching to playing -sending RECORD");
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_RECORD, 0);
ret = GST_STATE_CHANGE_SUCCESS;
break;
}
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* send pause request and keep the idle task around */
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_PAUSE,
CMD_LOOP);
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_CLOSE,
CMD_PAUSE);
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_rtsp_client_sink_stop (rtsp_client_sink);
ret = GST_STATE_CHANGE_SUCCESS;
break;
default:
break;
}
done:
return ret;
start_failed:
{
GST_DEBUG_OBJECT (rtsp_client_sink, "start failed");
return GST_STATE_CHANGE_FAILURE;
}
}
/*** GSTURIHANDLER INTERFACE *************************************************/
static GstURIType
gst_rtsp_client_sink_uri_get_type (GType type)
{
return GST_URI_SINK;
}
static const gchar *const *
gst_rtsp_client_sink_uri_get_protocols (GType type)
{
static const gchar *protocols[] =
{ "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
"rtsps", "rtspsu", "rtspst", "rtspsh", NULL
};
return protocols;
}
static gchar *
gst_rtsp_client_sink_uri_get_uri (GstURIHandler * handler)
{
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (handler);
/* FIXME: make thread-safe */
return g_strdup (sink->conninfo.location);
}
static gboolean
gst_rtsp_client_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri,
GError ** error)
{
GstRTSPClientSink *sink;
GstRTSPResult res;
GstSDPResult sres;
GstRTSPUrl *newurl = NULL;
GstSDPMessage *sdp = NULL;
sink = GST_RTSP_CLIENT_SINK (handler);
/* same URI, we're fine */
if (sink->conninfo.location && uri && !strcmp (uri, sink->conninfo.location))
goto was_ok;
if (g_str_has_prefix (uri, "rtsp-sdp://")) {
sres = gst_sdp_message_new (&sdp);
if (sres < 0)
goto sdp_failed;
GST_DEBUG_OBJECT (sink, "parsing SDP message");
sres = gst_sdp_message_parse_uri (uri, sdp);
if (sres < 0)
goto invalid_sdp;
} else {
/* try to parse */
GST_DEBUG_OBJECT (sink, "parsing URI");
if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
goto parse_error;
}
/* if worked, free previous and store new url object along with the original
* location. */
GST_DEBUG_OBJECT (sink, "configuring URI");
g_free (sink->conninfo.location);
sink->conninfo.location = g_strdup (uri);
gst_rtsp_url_free (sink->conninfo.url);
sink->conninfo.url = newurl;
g_free (sink->conninfo.url_str);
if (newurl)
sink->conninfo.url_str = gst_rtsp_url_get_request_uri (sink->conninfo.url);
else
sink->conninfo.url_str = NULL;
if (sink->uri_sdp)
gst_sdp_message_free (sink->uri_sdp);
sink->uri_sdp = sdp;
sink->from_sdp = sdp != NULL;
GST_DEBUG_OBJECT (sink, "set uri: %s", GST_STR_NULL (uri));
GST_DEBUG_OBJECT (sink, "request uri is: %s",
GST_STR_NULL (sink->conninfo.url_str));
return TRUE;
/* Special cases */
was_ok:
{
GST_DEBUG_OBJECT (sink, "URI was ok: '%s'", GST_STR_NULL (uri));
return TRUE;
}
sdp_failed:
{
GST_ERROR_OBJECT (sink, "Could not create new SDP (%d)", sres);
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
"Could not create SDP");
return FALSE;
}
invalid_sdp:
{
GST_ERROR_OBJECT (sink, "Not a valid SDP (%d) '%s'", sres,
GST_STR_NULL (uri));
gst_sdp_message_free (sdp);
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
"Invalid SDP");
return FALSE;
}
parse_error:
{
GST_ERROR_OBJECT (sink, "Not a valid RTSP url '%s' (%d)",
GST_STR_NULL (uri), res);
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
"Invalid RTSP URI");
return FALSE;
}
}
static void
gst_rtsp_client_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_rtsp_client_sink_uri_get_type;
iface->get_protocols = gst_rtsp_client_sink_uri_get_protocols;
iface->get_uri = gst_rtsp_client_sink_uri_get_uri;
iface->set_uri = gst_rtsp_client_sink_uri_set_uri;
}
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